diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
new file mode 100644
index 0000000..1ceb850
--- /dev/null
+++ b/services/audioflinger/Threads.cpp
@@ -0,0 +1,4426 @@
+/*
+**
+** Copyright 2012, 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 <fcntl.h>
+#include <sys/stat.h>
+#include <cutils/properties.h>
+#include <cutils/compiler.h>
+#include <utils/Log.h>
+
+#include <private/media/AudioTrackShared.h>
+#include <hardware/audio.h>
+#include <audio_effects/effect_ns.h>
+#include <audio_effects/effect_aec.h>
+#include <audio_utils/primitives.h>
+
+// NBAIO implementations
+#include <media/nbaio/AudioStreamOutSink.h>
+#include <media/nbaio/MonoPipe.h>
+#include <media/nbaio/MonoPipeReader.h>
+#include <media/nbaio/Pipe.h>
+#include <media/nbaio/PipeReader.h>
+#include <media/nbaio/SourceAudioBufferProvider.h>
+
+#include <powermanager/PowerManager.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
+
+#include "AudioFlinger.h"
+#include "AudioMixer.h"
+#include "FastMixer.h"
+#include "ServiceUtilities.h"
+#include "SchedulingPolicyService.h"
+
+#undef ADD_BATTERY_DATA
+
+#ifdef ADD_BATTERY_DATA
+#include <media/IMediaPlayerService.h>
+#include <media/IMediaDeathNotifier.h>
+#endif
+
+// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
+#ifdef DEBUG_CPU_USAGE
+#include <cpustats/CentralTendencyStatistics.h>
+#include <cpustats/ThreadCpuUsage.h>
+#endif
+
+// ----------------------------------------------------------------------------
+
+// Note: the following macro is used for extremely verbose logging message.  In
+// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
+// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
+// are so verbose that we want to suppress them even when we have ALOG_ASSERT
+// turned on.  Do not uncomment the #def below unless you really know what you
+// are doing and want to see all of the extremely verbose messages.
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+namespace android {
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+// allow less retry attempts on direct output thread.
+// direct outputs can be a scarce resource in audio hardware and should
+// be released as quickly as possible.
+static const int8_t kMaxTrackRetriesDirect = 2;
+
+// don't warn about blocked writes or record buffer overflows more often than this
+static const nsecs_t kWarningThrottleNs = seconds(5);
+
+// RecordThread loop sleep time upon application overrun or audio HAL read error
+static const int kRecordThreadSleepUs = 5000;
+
+// maximum time to wait for setParameters to complete
+static const nsecs_t kSetParametersTimeoutNs = seconds(2);
+
+// minimum sleep time for the mixer thread loop when tracks are active but in underrun
+static const uint32_t kMinThreadSleepTimeUs = 5000;
+// maximum divider applied to the active sleep time in the mixer thread loop
+static const uint32_t kMaxThreadSleepTimeShift = 2;
+
+// minimum normal mix buffer size, expressed in milliseconds rather than frames
+static const uint32_t kMinNormalMixBufferSizeMs = 20;
+// maximum normal mix buffer size
+static const uint32_t kMaxNormalMixBufferSizeMs = 24;
+
+// Whether to use fast mixer
+static const enum {
+    FastMixer_Never,    // never initialize or use: for debugging only
+    FastMixer_Always,   // always initialize and use, even if not needed: for debugging only
+                        // normal mixer multiplier is 1
+    FastMixer_Static,   // initialize if needed, then use all the time if initialized,
+                        // multiplier is calculated based on min & max normal mixer buffer size
+    FastMixer_Dynamic,  // initialize if needed, then use dynamically depending on track load,
+                        // multiplier is calculated based on min & max normal mixer buffer size
+    // FIXME for FastMixer_Dynamic:
+    //  Supporting this option will require fixing HALs that can't handle large writes.
+    //  For example, one HAL implementation returns an error from a large write,
+    //  and another HAL implementation corrupts memory, possibly in the sample rate converter.
+    //  We could either fix the HAL implementations, or provide a wrapper that breaks
+    //  up large writes into smaller ones, and the wrapper would need to deal with scheduler.
+} kUseFastMixer = FastMixer_Static;
+
+// Priorities for requestPriority
+static const int kPriorityAudioApp = 2;
+static const int kPriorityFastMixer = 3;
+
+// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
+// for the track.  The client then sub-divides this into smaller buffers for its use.
+// Currently the client uses double-buffering by default, but doesn't tell us about that.
+// So for now we just assume that client is double-buffered.
+// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
+// N-buffering, so AudioFlinger could allocate the right amount of memory.
+// See the client's minBufCount and mNotificationFramesAct calculations for details.
+static const int kFastTrackMultiplier = 2;
+
+// ----------------------------------------------------------------------------
+
+#ifdef ADD_BATTERY_DATA
+// To collect the amplifier usage
+static void addBatteryData(uint32_t params) {
+    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
+    if (service == NULL) {
+        // it already logged
+        return;
+    }
+
+    service->addBatteryData(params);
+}
+#endif
+
+
+// ----------------------------------------------------------------------------
+//      CPU Stats
+// ----------------------------------------------------------------------------
+
+class CpuStats {
+public:
+    CpuStats();
+    void sample(const String8 &title);
+#ifdef DEBUG_CPU_USAGE
+private:
+    ThreadCpuUsage mCpuUsage;           // instantaneous thread CPU usage in wall clock ns
+    CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
+
+    CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
+
+    int mCpuNum;                        // thread's current CPU number
+    int mCpukHz;                        // frequency of thread's current CPU in kHz
+#endif
+};
+
+CpuStats::CpuStats()
+#ifdef DEBUG_CPU_USAGE
+    : mCpuNum(-1), mCpukHz(-1)
+#endif
+{
+}
+
+void CpuStats::sample(const String8 &title) {
+#ifdef DEBUG_CPU_USAGE
+    // get current thread's delta CPU time in wall clock ns
+    double wcNs;
+    bool valid = mCpuUsage.sampleAndEnable(wcNs);
+
+    // record sample for wall clock statistics
+    if (valid) {
+        mWcStats.sample(wcNs);
+    }
+
+    // get the current CPU number
+    int cpuNum = sched_getcpu();
+
+    // get the current CPU frequency in kHz
+    int cpukHz = mCpuUsage.getCpukHz(cpuNum);
+
+    // check if either CPU number or frequency changed
+    if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
+        mCpuNum = cpuNum;
+        mCpukHz = cpukHz;
+        // ignore sample for purposes of cycles
+        valid = false;
+    }
+
+    // if no change in CPU number or frequency, then record sample for cycle statistics
+    if (valid && mCpukHz > 0) {
+        double cycles = wcNs * cpukHz * 0.000001;
+        mHzStats.sample(cycles);
+    }
+
+    unsigned n = mWcStats.n();
+    // mCpuUsage.elapsed() is expensive, so don't call it every loop
+    if ((n & 127) == 1) {
+        long long elapsed = mCpuUsage.elapsed();
+        if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
+            double perLoop = elapsed / (double) n;
+            double perLoop100 = perLoop * 0.01;
+            double perLoop1k = perLoop * 0.001;
+            double mean = mWcStats.mean();
+            double stddev = mWcStats.stddev();
+            double minimum = mWcStats.minimum();
+            double maximum = mWcStats.maximum();
+            double meanCycles = mHzStats.mean();
+            double stddevCycles = mHzStats.stddev();
+            double minCycles = mHzStats.minimum();
+            double maxCycles = mHzStats.maximum();
+            mCpuUsage.resetElapsed();
+            mWcStats.reset();
+            mHzStats.reset();
+            ALOGD("CPU usage for %s over past %.1f secs\n"
+                "  (%u mixer loops at %.1f mean ms per loop):\n"
+                "  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
+                "  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
+                "  MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
+                    title.string(),
+                    elapsed * .000000001, n, perLoop * .000001,
+                    mean * .001,
+                    stddev * .001,
+                    minimum * .001,
+                    maximum * .001,
+                    mean / perLoop100,
+                    stddev / perLoop100,
+                    minimum / perLoop100,
+                    maximum / perLoop100,
+                    meanCycles / perLoop1k,
+                    stddevCycles / perLoop1k,
+                    minCycles / perLoop1k,
+                    maxCycles / perLoop1k);
+
+        }
+    }
+#endif
+};
+
+// ----------------------------------------------------------------------------
+//      ThreadBase
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+        audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
+    :   Thread(false /*canCallJava*/),
+        mType(type),
+        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
+        // mChannelMask
+        mChannelCount(0),
+        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+        mParamStatus(NO_ERROR),
+        mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
+        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
+        // mName will be set by concrete (non-virtual) subclass
+        mDeathRecipient(new PMDeathRecipient(this))
+{
+}
+
+AudioFlinger::ThreadBase::~ThreadBase()
+{
+    mParamCond.broadcast();
+    // do not lock the mutex in destructor
+    releaseWakeLock_l();
+    if (mPowerManager != 0) {
+        sp<IBinder> binder = mPowerManager->asBinder();
+        binder->unlinkToDeath(mDeathRecipient);
+    }
+}
+
+void AudioFlinger::ThreadBase::exit()
+{
+    ALOGV("ThreadBase::exit");
+    // do any cleanup required for exit to succeed
+    preExit();
+    {
+        // This lock prevents the following race in thread (uniprocessor for illustration):
+        //  if (!exitPending()) {
+        //      // context switch from here to exit()
+        //      // exit() calls requestExit(), what exitPending() observes
+        //      // exit() calls signal(), which is dropped since no waiters
+        //      // context switch back from exit() to here
+        //      mWaitWorkCV.wait(...);
+        //      // now thread is hung
+        //  }
+        AutoMutex lock(mLock);
+        requestExit();
+        mWaitWorkCV.broadcast();
+    }
+    // When Thread::requestExitAndWait is made virtual and this method is renamed to
+    // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
+    requestExitAndWait();
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+    status_t status;
+
+    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
+    Mutex::Autolock _l(mLock);
+
+    mNewParameters.add(keyValuePairs);
+    mWaitWorkCV.signal();
+    // wait condition with timeout in case the thread loop has exited
+    // before the request could be processed
+    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
+        status = mParamStatus;
+        mWaitWorkCV.signal();
+    } else {
+        status = TIMED_OUT;
+    }
+    return status;
+}
+
+void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
+{
+    Mutex::Autolock _l(mLock);
+    sendIoConfigEvent_l(event, param);
+}
+
+// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param)
+{
+    IoConfigEvent *ioEvent = new IoConfigEvent(event, param);
+    mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent));
+    ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event,
+            param);
+    mWaitWorkCV.signal();
+}
+
+// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
+{
+    PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio);
+    mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent));
+    ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d",
+          mConfigEvents.size(), pid, tid, prio);
+    mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+    mLock.lock();
+    while (!mConfigEvents.isEmpty()) {
+        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+        ConfigEvent *event = mConfigEvents[0];
+        mConfigEvents.removeAt(0);
+        // release mLock before locking AudioFlinger mLock: lock order is always
+        // AudioFlinger then ThreadBase to avoid cross deadlock
+        mLock.unlock();
+        switch(event->type()) {
+            case CFG_EVENT_PRIO: {
+                PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
+                int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio());
+                if (err != 0) {
+                    ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
+                          "error %d",
+                          prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
+                }
+            } break;
+            case CFG_EVENT_IO: {
+                IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
+                mAudioFlinger->mLock.lock();
+                audioConfigChanged_l(ioEvent->event(), ioEvent->param());
+                mAudioFlinger->mLock.unlock();
+            } break;
+            default:
+                ALOGE("processConfigEvents() unknown event type %d", event->type());
+                break;
+        }
+        delete event;
+        mLock.lock();
+    }
+    mLock.unlock();
+}
+
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = AudioFlinger::dumpTryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
+        write(fd, buffer, strlen(buffer));
+    }
+
+    snprintf(buffer, SIZE, "io handle: %d\n", mId);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "TID: %d\n", getTid());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
+    result.append(buffer);
+    result.append(" Index Command");
+    for (size_t i = 0; i < mNewParameters.size(); ++i) {
+        snprintf(buffer, SIZE, "\n %02d    ", i);
+        result.append(buffer);
+        result.append(mNewParameters[i]);
+    }
+
+    snprintf(buffer, SIZE, "\n\nPending config events: \n");
+    result.append(buffer);
+    for (size_t i = 0; i < mConfigEvents.size(); i++) {
+        mConfigEvents[i]->dump(buffer, SIZE);
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+
+    if (locked) {
+        mLock.unlock();
+    }
+}
+
+void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+    write(fd, buffer, strlen(buffer));
+
+    for (size_t i = 0; i < mEffectChains.size(); ++i) {
+        sp<EffectChain> chain = mEffectChains[i];
+        if (chain != 0) {
+            chain->dump(fd, args);
+        }
+    }
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock()
+{
+    Mutex::Autolock _l(mLock);
+    acquireWakeLock_l();
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock_l()
+{
+    if (mPowerManager == 0) {
+        // use checkService() to avoid blocking if power service is not up yet
+        sp<IBinder> binder =
+            defaultServiceManager()->checkService(String16("power"));
+        if (binder == 0) {
+            ALOGW("Thread %s cannot connect to the power manager service", mName);
+        } else {
+            mPowerManager = interface_cast<IPowerManager>(binder);
+            binder->linkToDeath(mDeathRecipient);
+        }
+    }
+    if (mPowerManager != 0) {
+        sp<IBinder> binder = new BBinder();
+        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
+                                                         binder,
+                                                         String16(mName));
+        if (status == NO_ERROR) {
+            mWakeLockToken = binder;
+        }
+        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+    }
+}
+
+void AudioFlinger::ThreadBase::releaseWakeLock()
+{
+    Mutex::Autolock _l(mLock);
+    releaseWakeLock_l();
+}
+
+void AudioFlinger::ThreadBase::releaseWakeLock_l()
+{
+    if (mWakeLockToken != 0) {
+        ALOGV("releaseWakeLock_l() %s", mName);
+        if (mPowerManager != 0) {
+            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
+        }
+        mWakeLockToken.clear();
+    }
+}
+
+void AudioFlinger::ThreadBase::clearPowerManager()
+{
+    Mutex::Autolock _l(mLock);
+    releaseWakeLock_l();
+    mPowerManager.clear();
+}
+
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        thread->clearPowerManager();
+    }
+    ALOGW("power manager service died !!!");
+}
+
+void AudioFlinger::ThreadBase::setEffectSuspended(
+        const effect_uuid_t *type, bool suspend, int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    setEffectSuspended_l(type, suspend, sessionId);
+}
+
+void AudioFlinger::ThreadBase::setEffectSuspended_l(
+        const effect_uuid_t *type, bool suspend, int sessionId)
+{
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    if (chain != 0) {
+        if (type != NULL) {
+            chain->setEffectSuspended_l(type, suspend);
+        } else {
+            chain->setEffectSuspendedAll_l(suspend);
+        }
+    }
+
+    updateSuspendedSessions_l(type, suspend, sessionId);
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
+{
+    ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
+    if (index < 0) {
+        return;
+    }
+
+    const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
+            mSuspendedSessions.valueAt(index);
+
+    for (size_t i = 0; i < sessionEffects.size(); i++) {
+        sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
+        for (int j = 0; j < desc->mRefCount; j++) {
+            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
+                chain->setEffectSuspendedAll_l(true);
+            } else {
+                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
+                    desc->mType.timeLow);
+                chain->setEffectSuspended_l(&desc->mType, true);
+            }
+        }
+    }
+}
+
+void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
+                                                         bool suspend,
+                                                         int sessionId)
+{
+    ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
+
+    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
+
+    if (suspend) {
+        if (index >= 0) {
+            sessionEffects = mSuspendedSessions.valueAt(index);
+        } else {
+            mSuspendedSessions.add(sessionId, sessionEffects);
+        }
+    } else {
+        if (index < 0) {
+            return;
+        }
+        sessionEffects = mSuspendedSessions.valueAt(index);
+    }
+
+
+    int key = EffectChain::kKeyForSuspendAll;
+    if (type != NULL) {
+        key = type->timeLow;
+    }
+    index = sessionEffects.indexOfKey(key);
+
+    sp<SuspendedSessionDesc> desc;
+    if (suspend) {
+        if (index >= 0) {
+            desc = sessionEffects.valueAt(index);
+        } else {
+            desc = new SuspendedSessionDesc();
+            if (type != NULL) {
+                desc->mType = *type;
+            }
+            sessionEffects.add(key, desc);
+            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
+        }
+        desc->mRefCount++;
+    } else {
+        if (index < 0) {
+            return;
+        }
+        desc = sessionEffects.valueAt(index);
+        if (--desc->mRefCount == 0) {
+            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
+            sessionEffects.removeItemsAt(index);
+            if (sessionEffects.isEmpty()) {
+                ALOGV("updateSuspendedSessions_l() restore removing session %d",
+                                 sessionId);
+                mSuspendedSessions.removeItem(sessionId);
+            }
+        }
+    }
+    if (!sessionEffects.isEmpty()) {
+        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
+    }
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
+                                                            bool enabled,
+                                                            int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
+                                                            bool enabled,
+                                                            int sessionId)
+{
+    if (mType != RECORD) {
+        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
+        // another session. This gives the priority to well behaved effect control panels
+        // and applications not using global effects.
+        // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
+        // global effects
+        if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
+            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
+        }
+    }
+
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    if (chain != 0) {
+        chain->checkSuspendOnEffectEnabled(effect, enabled);
+    }
+}
+
+// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
+        const sp<AudioFlinger::Client>& client,
+        const sp<IEffectClient>& effectClient,
+        int32_t priority,
+        int sessionId,
+        effect_descriptor_t *desc,
+        int *enabled,
+        status_t *status
+        )
+{
+    sp<EffectModule> effect;
+    sp<EffectHandle> handle;
+    status_t lStatus;
+    sp<EffectChain> chain;
+    bool chainCreated = false;
+    bool effectCreated = false;
+    bool effectRegistered = false;
+
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGW("createEffect_l() Audio driver not initialized.");
+        goto Exit;
+    }
+
+    // Do not allow effects with session ID 0 on direct output or duplicating threads
+    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
+    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
+        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
+                desc->name, sessionId);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+    // Only Pre processor effects are allowed on input threads and only on input threads
+    if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
+        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
+                desc->name, desc->flags, mType);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        // check for existing effect chain with the requested audio session
+        chain = getEffectChain_l(sessionId);
+        if (chain == 0) {
+            // create a new chain for this session
+            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
+            chain = new EffectChain(this, sessionId);
+            addEffectChain_l(chain);
+            chain->setStrategy(getStrategyForSession_l(sessionId));
+            chainCreated = true;
+        } else {
+            effect = chain->getEffectFromDesc_l(desc);
+        }
+
+        ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
+
+        if (effect == 0) {
+            int id = mAudioFlinger->nextUniqueId();
+            // Check CPU and memory usage
+            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            effectRegistered = true;
+            // create a new effect module if none present in the chain
+            effect = new EffectModule(this, chain, desc, id, sessionId);
+            lStatus = effect->status();
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            lStatus = chain->addEffect_l(effect);
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            effectCreated = true;
+
+            effect->setDevice(mOutDevice);
+            effect->setDevice(mInDevice);
+            effect->setMode(mAudioFlinger->getMode());
+            effect->setAudioSource(mAudioSource);
+        }
+        // create effect handle and connect it to effect module
+        handle = new EffectHandle(effect, client, effectClient, priority);
+        lStatus = effect->addHandle(handle.get());
+        if (enabled != NULL) {
+            *enabled = (int)effect->isEnabled();
+        }
+    }
+
+Exit:
+    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
+        Mutex::Autolock _l(mLock);
+        if (effectCreated) {
+            chain->removeEffect_l(effect);
+        }
+        if (effectRegistered) {
+            AudioSystem::unregisterEffect(effect->id());
+        }
+        if (chainCreated) {
+            removeEffectChain_l(chain);
+        }
+        handle.clear();
+    }
+
+    if (status != NULL) {
+        *status = lStatus;
+    }
+    return handle;
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
+{
+    Mutex::Autolock _l(mLock);
+    return getEffect_l(sessionId, effectId);
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
+{
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
+}
+
+// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
+// PlaybackThread::mLock held
+status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
+{
+    // check for existing effect chain with the requested audio session
+    int sessionId = effect->sessionId();
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    bool chainCreated = false;
+
+    if (chain == 0) {
+        // create a new chain for this session
+        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
+        chain = new EffectChain(this, sessionId);
+        addEffectChain_l(chain);
+        chain->setStrategy(getStrategyForSession_l(sessionId));
+        chainCreated = true;
+    }
+    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
+
+    if (chain->getEffectFromId_l(effect->id()) != 0) {
+        ALOGW("addEffect_l() %p effect %s already present in chain %p",
+                this, effect->desc().name, chain.get());
+        return BAD_VALUE;
+    }
+
+    status_t status = chain->addEffect_l(effect);
+    if (status != NO_ERROR) {
+        if (chainCreated) {
+            removeEffectChain_l(chain);
+        }
+        return status;
+    }
+
+    effect->setDevice(mOutDevice);
+    effect->setDevice(mInDevice);
+    effect->setMode(mAudioFlinger->getMode());
+    effect->setAudioSource(mAudioSource);
+    return NO_ERROR;
+}
+
+void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
+
+    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
+    effect_descriptor_t desc = effect->desc();
+    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+        detachAuxEffect_l(effect->id());
+    }
+
+    sp<EffectChain> chain = effect->chain().promote();
+    if (chain != 0) {
+        // remove effect chain if removing last effect
+        if (chain->removeEffect_l(effect) == 0) {
+            removeEffectChain_l(chain);
+        }
+    } else {
+        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
+    }
+}
+
+void AudioFlinger::ThreadBase::lockEffectChains_l(
+        Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+{
+    effectChains = mEffectChains;
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        mEffectChains[i]->lock();
+    }
+}
+
+void AudioFlinger::ThreadBase::unlockEffectChains(
+        const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+{
+    for (size_t i = 0; i < effectChains.size(); i++) {
+        effectChains[i]->unlock();
+    }
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    return getEffectChain_l(sessionId);
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
+{
+    size_t size = mEffectChains.size();
+    for (size_t i = 0; i < size; i++) {
+        if (mEffectChains[i]->sessionId() == sessionId) {
+            return mEffectChains[i];
+        }
+    }
+    return 0;
+}
+
+void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
+{
+    Mutex::Autolock _l(mLock);
+    size_t size = mEffectChains.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffectChains[i]->setMode_l(mode);
+    }
+}
+
+void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
+                                                    EffectHandle *handle,
+                                                    bool unpinIfLast) {
+
+    Mutex::Autolock _l(mLock);
+    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
+    // delete the effect module if removing last handle on it
+    if (effect->removeHandle(handle) == 0) {
+        if (!effect->isPinned() || unpinIfLast) {
+            removeEffect_l(effect);
+            AudioSystem::unregisterEffect(effect->id());
+        }
+    }
+}
+
+// ----------------------------------------------------------------------------
+//      Playback
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
+                                             AudioStreamOut* output,
+                                             audio_io_handle_t id,
+                                             audio_devices_t device,
+                                             type_t type)
+    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
+        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+        // mStreamTypes[] initialized in constructor body
+        mOutput(output),
+        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
+        mMixerStatus(MIXER_IDLE),
+        mMixerStatusIgnoringFastTracks(MIXER_IDLE),
+        standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
+        mScreenState(AudioFlinger::mScreenState),
+        // index 0 is reserved for normal mixer's submix
+        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
+{
+    snprintf(mName, kNameLength, "AudioOut_%X", id);
+
+    // Assumes constructor is called by AudioFlinger with it's mLock held, but
+    // it would be safer to explicitly pass initial masterVolume/masterMute as
+    // parameter.
+    //
+    // If the HAL we are using has support for master volume or master mute,
+    // then do not attenuate or mute during mixing (just leave the volume at 1.0
+    // and the mute set to false).
+    mMasterVolume = audioFlinger->masterVolume_l();
+    mMasterMute = audioFlinger->masterMute_l();
+    if (mOutput && mOutput->audioHwDev) {
+        if (mOutput->audioHwDev->canSetMasterVolume()) {
+            mMasterVolume = 1.0;
+        }
+
+        if (mOutput->audioHwDev->canSetMasterMute()) {
+            mMasterMute = false;
+        }
+    }
+
+    readOutputParameters();
+
+    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
+    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
+    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
+            stream = (audio_stream_type_t) (stream + 1)) {
+        mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
+        mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
+    }
+    // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
+    // because mAudioFlinger doesn't have one to copy from
+}
+
+AudioFlinger::PlaybackThread::~PlaybackThread()
+{
+    delete [] mMixBuffer;
+}
+
+void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    dumpTracks(fd, args);
+    dumpEffectChains(fd, args);
+}
+
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    result.appendFormat("Output thread %p stream volumes in dB:\n    ", this);
+    for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
+        const stream_type_t *st = &mStreamTypes[i];
+        if (i > 0) {
+            result.appendFormat(", ");
+        }
+        result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
+        if (st->mute) {
+            result.append("M");
+        }
+    }
+    result.append("\n");
+    write(fd, result.string(), result.length());
+    result.clear();
+
+    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
+    result.append(buffer);
+    Track::appendDumpHeader(result);
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
+    result.append(buffer);
+    Track::appendDumpHeader(result);
+    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+        sp<Track> track = mActiveTracks[i].promote();
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+    write(fd, result.string(), result.size());
+
+    // These values are "raw"; they will wrap around.  See prepareTracks_l() for a better way.
+    FastTrackUnderruns underruns = getFastTrackUnderruns(0);
+    fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
+            underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
+}
+
+void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
+    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, "suspend count: %d\n", mSuspended);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
+
+    dumpBase(fd, args);
+}
+
+// Thread virtuals
+status_t AudioFlinger::PlaybackThread::readyToRun()
+{
+    status_t status = initCheck();
+    if (status == NO_ERROR) {
+        ALOGI("AudioFlinger's thread %p ready to run", this);
+    } else {
+        ALOGE("No working audio driver found.");
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::onFirstRef()
+{
+    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// ThreadBase virtuals
+void AudioFlinger::PlaybackThread::preExit()
+{
+    ALOGV("  preExit()");
+    // FIXME this is using hard-coded strings but in the future, this functionality will be
+    //       converted to use audio HAL extensions required to support tunneling
+    mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
+}
+
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+        const sp<AudioFlinger::Client>& client,
+        audio_stream_type_t streamType,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        const sp<IMemory>& sharedBuffer,
+        int sessionId,
+        IAudioFlinger::track_flags_t *flags,
+        pid_t tid,
+        status_t *status)
+{
+    sp<Track> track;
+    status_t lStatus;
+
+    bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
+
+    // client expresses a preference for FAST, but we get the final say
+    if (*flags & IAudioFlinger::TRACK_FAST) {
+      if (
+            // not timed
+            (!isTimed) &&
+            // either of these use cases:
+            (
+              // use case 1: shared buffer with any frame count
+              (
+                (sharedBuffer != 0)
+              ) ||
+              // use case 2: callback handler and frame count is default or at least as large as HAL
+              (
+                (tid != -1) &&
+                ((frameCount == 0) ||
+                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+              )
+            ) &&
+            // PCM data
+            audio_is_linear_pcm(format) &&
+            // mono or stereo
+            ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
+              (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
+#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
+            // hardware sample rate
+            (sampleRate == mSampleRate) &&
+#endif
+            // normal mixer has an associated fast mixer
+            hasFastMixer() &&
+            // there are sufficient fast track slots available
+            (mFastTrackAvailMask != 0)
+            // FIXME test that MixerThread for this fast track has a capable output HAL
+            // FIXME add a permission test also?
+        ) {
+        // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
+        if (frameCount == 0) {
+            frameCount = mFrameCount * kFastTrackMultiplier;
+        }
+        ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
+                frameCount, mFrameCount);
+      } else {
+        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
+                "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
+                "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
+                isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
+                audio_is_linear_pcm(format),
+                channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
+        *flags &= ~IAudioFlinger::TRACK_FAST;
+        // For compatibility with AudioTrack calculation, buffer depth is forced
+        // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
+        // This is probably too conservative, but legacy application code may depend on it.
+        // If you change this calculation, also review the start threshold which is related.
+        uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
+        uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
+        if (minBufCount < 2) {
+            minBufCount = 2;
+        }
+        size_t minFrameCount = mNormalFrameCount * minBufCount;
+        if (frameCount < minFrameCount) {
+            frameCount = minFrameCount;
+        }
+      }
+    }
+
+    if (mType == DIRECT) {
+        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
+            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
+                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
+                        "for output %p with format %d",
+                        sampleRate, format, channelMask, mOutput, mFormat);
+                lStatus = BAD_VALUE;
+                goto Exit;
+            }
+        }
+    } else {
+        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+        if (sampleRate > mSampleRate*2) {
+            ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+    }
+
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGE("Audio driver not initialized.");
+        goto Exit;
+    }
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        // all tracks in same audio session must share the same routing strategy otherwise
+        // conflicts will happen when tracks are moved from one output to another by audio policy
+        // manager
+        uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            sp<Track> t = mTracks[i];
+            if (t != 0 && !t->isOutputTrack()) {
+                uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
+                if (sessionId == t->sessionId() && strategy != actual) {
+                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
+                            strategy, actual);
+                    lStatus = BAD_VALUE;
+                    goto Exit;
+                }
+            }
+        }
+
+        if (!isTimed) {
+            track = new Track(this, client, streamType, sampleRate, format,
+                    channelMask, frameCount, sharedBuffer, sessionId, *flags);
+        } else {
+            track = TimedTrack::create(this, client, streamType, sampleRate, format,
+                    channelMask, frameCount, sharedBuffer, sessionId);
+        }
+        if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
+
+        sp<EffectChain> chain = getEffectChain_l(sessionId);
+        if (chain != 0) {
+            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
+            track->setMainBuffer(chain->inBuffer());
+            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
+            chain->incTrackCnt();
+        }
+
+        if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
+            pid_t callingPid = IPCThreadState::self()->getCallingPid();
+            // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
+            // so ask activity manager to do this on our behalf
+            sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
+        }
+    }
+
+    lStatus = NO_ERROR;
+
+Exit:
+    if (status) {
+        *status = lStatus;
+    }
+    return track;
+}
+
+uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
+{
+    return latency;
+}
+
+uint32_t AudioFlinger::PlaybackThread::latency() const
+{
+    Mutex::Autolock _l(mLock);
+    return latency_l();
+}
+uint32_t AudioFlinger::PlaybackThread::latency_l() const
+{
+    if (initCheck() == NO_ERROR) {
+        return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
+    } else {
+        return 0;
+    }
+}
+
+void AudioFlinger::PlaybackThread::setMasterVolume(float value)
+{
+    Mutex::Autolock _l(mLock);
+    // Don't apply master volume in SW if our HAL can do it for us.
+    if (mOutput && mOutput->audioHwDev &&
+        mOutput->audioHwDev->canSetMasterVolume()) {
+        mMasterVolume = 1.0;
+    } else {
+        mMasterVolume = value;
+    }
+}
+
+void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+{
+    Mutex::Autolock _l(mLock);
+    // Don't apply master mute in SW if our HAL can do it for us.
+    if (mOutput && mOutput->audioHwDev &&
+        mOutput->audioHwDev->canSetMasterMute()) {
+        mMasterMute = false;
+    } else {
+        mMasterMute = muted;
+    }
+}
+
+void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+{
+    Mutex::Autolock _l(mLock);
+    mStreamTypes[stream].volume = value;
+}
+
+void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+{
+    Mutex::Autolock _l(mLock);
+    mStreamTypes[stream].mute = muted;
+}
+
+float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
+{
+    Mutex::Autolock _l(mLock);
+    return mStreamTypes[stream].volume;
+}
+
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+{
+    status_t status = ALREADY_EXISTS;
+
+    // set retry count for buffer fill
+    track->mRetryCount = kMaxTrackStartupRetries;
+    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;
+        track->mResetDone = false;
+        track->mPresentationCompleteFrames = 0;
+        mActiveTracks.add(track);
+        if (track->mainBuffer() != mMixBuffer) {
+            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+            if (chain != 0) {
+                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
+                        track->sessionId());
+                chain->incActiveTrackCnt();
+            }
+        }
+
+        status = NO_ERROR;
+    }
+
+    ALOGV("mWaitWorkCV.broadcast");
+    mWaitWorkCV.broadcast();
+
+    return status;
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+{
+    track->mState = TrackBase::TERMINATED;
+    // active tracks are removed by threadLoop()
+    if (mActiveTracks.indexOf(track) < 0) {
+        removeTrack_l(track);
+    }
+}
+
+void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
+{
+    track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+    mTracks.remove(track);
+    deleteTrackName_l(track->name());
+    // redundant as track is about to be destroyed, for dumpsys only
+    track->mName = -1;
+    if (track->isFastTrack()) {
+        int index = track->mFastIndex;
+        ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
+        ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
+        mFastTrackAvailMask |= 1 << index;
+        // redundant as track is about to be destroyed, for dumpsys only
+        track->mFastIndex = -1;
+    }
+    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+    if (chain != 0) {
+        chain->decTrackCnt();
+    }
+}
+
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+{
+    String8 out_s8 = String8("");
+    char *s;
+
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return out_s8;
+    }
+
+    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
+    out_s8 = String8(s);
+    free(s);
+    return out_s8;
+}
+
+// audioConfigChanged_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = NULL;
+
+    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event,
+            param);
+
+    switch (event) {
+    case AudioSystem::OUTPUT_OPENED:
+    case AudioSystem::OUTPUT_CONFIG_CHANGED:
+        desc.channels = mChannelMask;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mNormalFrameCount; // FIXME see
+                                             // AudioFlinger::frameCount(audio_io_handle_t)
+        desc.latency = latency();
+        param2 = &desc;
+        break;
+
+    case AudioSystem::STREAM_CONFIG_CHANGED:
+        param2 = &param;
+    case AudioSystem::OUTPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
+    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
+    mChannelCount = (uint16_t)popcount(mChannelMask);
+    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
+    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
+    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
+    if (mFrameCount & 15) {
+        ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
+                mFrameCount);
+    }
+
+    // Calculate size of normal mix buffer relative to the HAL output buffer size
+    double multiplier = 1.0;
+    if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
+            kUseFastMixer == FastMixer_Dynamic)) {
+        size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
+        size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
+        // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
+        minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
+        maxNormalFrameCount = maxNormalFrameCount & ~15;
+        if (maxNormalFrameCount < minNormalFrameCount) {
+            maxNormalFrameCount = minNormalFrameCount;
+        }
+        multiplier = (double) minNormalFrameCount / (double) mFrameCount;
+        if (multiplier <= 1.0) {
+            multiplier = 1.0;
+        } else if (multiplier <= 2.0) {
+            if (2 * mFrameCount <= maxNormalFrameCount) {
+                multiplier = 2.0;
+            } else {
+                multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
+            }
+        } else {
+            // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
+            // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
+            // track, but we sometimes have to do this to satisfy the maximum frame count
+            // constraint)
+            // FIXME this rounding up should not be done if no HAL SRC
+            uint32_t truncMult = (uint32_t) multiplier;
+            if ((truncMult & 1)) {
+                if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
+                    ++truncMult;
+                }
+            }
+            multiplier = (double) truncMult;
+        }
+    }
+    mNormalFrameCount = multiplier * mFrameCount;
+    // round up to nearest 16 frames to satisfy AudioMixer
+    mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
+            mNormalFrameCount);
+
+    delete[] mMixBuffer;
+    mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
+    memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+
+    // force reconfiguration of effect chains and engines to take new buffer size and audio
+    // parameters into account
+    // Note that mLock is not held when readOutputParameters() is called from the constructor
+    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
+    // matter.
+    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
+    Vector< sp<EffectChain> > effectChains = mEffectChains;
+    for (size_t i = 0; i < effectChains.size(); i ++) {
+        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
+    }
+}
+
+
+status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
+{
+    if (halFrames == NULL || dspFrames == NULL) {
+        return BAD_VALUE;
+    }
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return INVALID_OPERATION;
+    }
+    size_t framesWritten = mBytesWritten / mFrameSize;
+    *halFrames = framesWritten;
+
+    if (isSuspended()) {
+        // return an estimation of rendered frames when the output is suspended
+        size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
+        *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
+        return NO_ERROR;
+    } else {
+        return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
+    }
+}
+
+uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
+{
+    Mutex::Autolock _l(mLock);
+    uint32_t result = 0;
+    if (getEffectChain_l(sessionId) != 0) {
+        result = EFFECT_SESSION;
+    }
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (sessionId == track->sessionId() &&
+                !(track->mCblk->flags & CBLK_INVALID)) {
+            result |= TRACK_SESSION;
+            break;
+        }
+    }
+
+    return result;
+}
+
+uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
+{
+    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
+    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
+    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
+        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+    }
+    for (size_t i = 0; i < mTracks.size(); i++) {
+        sp<Track> track = mTracks[i];
+        if (sessionId == track->sessionId() &&
+                !(track->mCblk->flags & CBLK_INVALID)) {
+            return AudioSystem::getStrategyForStream(track->streamType());
+        }
+    }
+    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+}
+
+
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+{
+    Mutex::Autolock _l(mLock);
+    return mOutput;
+}
+
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+{
+    Mutex::Autolock _l(mLock);
+    AudioStreamOut *output = mOutput;
+    mOutput = NULL;
+    // FIXME FastMixer might also have a raw ptr to mOutputSink;
+    //       must push a NULL and wait for ack
+    mOutputSink.clear();
+    mPipeSink.clear();
+    mNormalSink.clear();
+    return output;
+}
+
+// this method must always be called either with ThreadBase mLock held or inside the thread loop
+audio_stream_t* AudioFlinger::PlaybackThread::stream() const
+{
+    if (mOutput == NULL) {
+        return NULL;
+    }
+    return &mOutput->stream->common;
+}
+
+uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
+{
+    return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
+}
+
+status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
+{
+    if (!isValidSyncEvent(event)) {
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock _l(mLock);
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (event->triggerSession() == track->sessionId()) {
+            (void) track->setSyncEvent(event);
+            return NO_ERROR;
+        }
+    }
+
+    return NAME_NOT_FOUND;
+}
+
+bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+{
+    return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
+}
+
+void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
+        const Vector< sp<Track> >& tracksToRemove)
+{
+    size_t count = tracksToRemove.size();
+    if (CC_UNLIKELY(count)) {
+        for (size_t i = 0 ; i < count ; i++) {
+            const sp<Track>& track = tracksToRemove.itemAt(i);
+            if ((track->sharedBuffer() != 0) &&
+                    (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) {
+                AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
+            }
+        }
+    }
+
+}
+
+void AudioFlinger::PlaybackThread::checkSilentMode_l()
+{
+    if (!mMasterMute) {
+        char value[PROPERTY_VALUE_MAX];
+        if (property_get("ro.audio.silent", value, "0") > 0) {
+            char *endptr;
+            unsigned long ul = strtoul(value, &endptr, 0);
+            if (*endptr == '\0' && ul != 0) {
+                ALOGD("Silence is golden");
+                // The setprop command will not allow a property to be changed after
+                // the first time it is set, so we don't have to worry about un-muting.
+                setMasterMute_l(true);
+            }
+        }
+    }
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_write()
+{
+    // FIXME rewrite to reduce number of system calls
+    mLastWriteTime = systemTime();
+    mInWrite = true;
+    int bytesWritten;
+
+    // If an NBAIO sink is present, use it to write the normal mixer's submix
+    if (mNormalSink != 0) {
+#define mBitShift 2 // FIXME
+        size_t count = mixBufferSize >> mBitShift;
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+        Tracer::traceBegin(ATRACE_TAG, "write");
+#endif
+        // update the setpoint when AudioFlinger::mScreenState changes
+        uint32_t screenState = AudioFlinger::mScreenState;
+        if (screenState != mScreenState) {
+            mScreenState = screenState;
+            MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+            if (pipe != NULL) {
+                pipe->setAvgFrames((mScreenState & 1) ?
+                        (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+            }
+        }
+        ssize_t framesWritten = mNormalSink->write(mMixBuffer, count);
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+        Tracer::traceEnd(ATRACE_TAG);
+#endif
+        if (framesWritten > 0) {
+            bytesWritten = framesWritten << mBitShift;
+        } else {
+            bytesWritten = framesWritten;
+        }
+    // otherwise use the HAL / AudioStreamOut directly
+    } else {
+        // Direct output thread.
+        bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
+    }
+
+    if (bytesWritten > 0) {
+        mBytesWritten += mixBufferSize;
+    }
+    mNumWrites++;
+    mInWrite = false;
+}
+
+/*
+The derived values that are cached:
+ - mixBufferSize from frame count * frame size
+ - activeSleepTime from activeSleepTimeUs()
+ - idleSleepTime from idleSleepTimeUs()
+ - standbyDelay from mActiveSleepTimeUs (DIRECT only)
+ - maxPeriod from frame count and sample rate (MIXER only)
+
+The parameters that affect these derived values are:
+ - frame count
+ - frame size
+ - sample rate
+ - device type: A2DP or not
+ - device latency
+ - format: PCM or not
+ - active sleep time
+ - idle sleep time
+*/
+
+void AudioFlinger::PlaybackThread::cacheParameters_l()
+{
+    mixBufferSize = mNormalFrameCount * mFrameSize;
+    activeSleepTime = activeSleepTimeUs();
+    idleSleepTime = idleSleepTimeUs();
+}
+
+void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+{
+    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
+            this,  streamType, mTracks.size());
+    Mutex::Autolock _l(mLock);
+
+    size_t size = mTracks.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = mTracks[i];
+        if (t->streamType() == streamType) {
+            android_atomic_or(CBLK_INVALID, &t->mCblk->flags);
+            t->mCblk->cv.signal();
+        }
+    }
+}
+
+status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+    int session = chain->sessionId();
+    int16_t *buffer = mMixBuffer;
+    bool ownsBuffer = false;
+
+    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
+    if (session > 0) {
+        // Only one effect chain can be present in direct output thread and it uses
+        // the mix buffer as input
+        if (mType != DIRECT) {
+            size_t numSamples = mNormalFrameCount * mChannelCount;
+            buffer = new int16_t[numSamples];
+            memset(buffer, 0, numSamples * sizeof(int16_t));
+            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
+            ownsBuffer = true;
+        }
+
+        // Attach all tracks with same session ID to this chain.
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            sp<Track> track = mTracks[i];
+            if (session == track->sessionId()) {
+                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
+                        buffer);
+                track->setMainBuffer(buffer);
+                chain->incTrackCnt();
+            }
+        }
+
+        // indicate all active tracks in the chain
+        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+            sp<Track> track = mActiveTracks[i].promote();
+            if (track == 0) {
+                continue;
+            }
+            if (session == track->sessionId()) {
+                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
+                chain->incActiveTrackCnt();
+            }
+        }
+    }
+
+    chain->setInBuffer(buffer, ownsBuffer);
+    chain->setOutBuffer(mMixBuffer);
+    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
+    // chains list in order to be processed last as it contains output stage effects
+    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
+    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
+    // after track specific effects and before output stage
+    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
+    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
+    // Effect chain for other sessions are inserted at beginning of effect
+    // chains list to be processed before output mix effects. Relative order between other
+    // sessions is not important
+    size_t size = mEffectChains.size();
+    size_t i = 0;
+    for (i = 0; i < size; i++) {
+        if (mEffectChains[i]->sessionId() < session) {
+            break;
+        }
+    }
+    mEffectChains.insertAt(chain, i);
+    checkSuspendOnAddEffectChain_l(chain);
+
+    return NO_ERROR;
+}
+
+size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+    int session = chain->sessionId();
+
+    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
+
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        if (chain == mEffectChains[i]) {
+            mEffectChains.removeAt(i);
+            // detach all active tracks from the chain
+            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+                sp<Track> track = mActiveTracks[i].promote();
+                if (track == 0) {
+                    continue;
+                }
+                if (session == track->sessionId()) {
+                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
+                            chain.get(), session);
+                    chain->decActiveTrackCnt();
+                }
+            }
+
+            // detach all tracks with same session ID from this chain
+            for (size_t i = 0; i < mTracks.size(); ++i) {
+                sp<Track> track = mTracks[i];
+                if (session == track->sessionId()) {
+                    track->setMainBuffer(mMixBuffer);
+                    chain->decTrackCnt();
+                }
+            }
+            break;
+        }
+    }
+    return mEffectChains.size();
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect(
+        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+    Mutex::Autolock _l(mLock);
+    return attachAuxEffect_l(track, EffectId);
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
+        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+    status_t status = NO_ERROR;
+
+    if (EffectId == 0) {
+        track->setAuxBuffer(0, NULL);
+    } else {
+        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
+        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+        if (effect != 0) {
+            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
+            } else {
+                status = INVALID_OPERATION;
+            }
+        } else {
+            status = BAD_VALUE;
+        }
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
+{
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (track->auxEffectId() == effectId) {
+            attachAuxEffect_l(track, 0);
+        }
+    }
+}
+
+bool AudioFlinger::PlaybackThread::threadLoop()
+{
+    Vector< sp<Track> > tracksToRemove;
+
+    standbyTime = systemTime();
+
+    // MIXER
+    nsecs_t lastWarning = 0;
+
+    // DUPLICATING
+    // FIXME could this be made local to while loop?
+    writeFrames = 0;
+
+    cacheParameters_l();
+    sleepTime = idleSleepTime;
+
+    if (mType == MIXER) {
+        sleepTimeShift = 0;
+    }
+
+    CpuStats cpuStats;
+    const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
+
+    acquireWakeLock();
+
+    while (!exitPending())
+    {
+        cpuStats.sample(myName);
+
+        Vector< sp<EffectChain> > effectChains;
+
+        processConfigEvents();
+
+        { // scope for mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                cacheParameters_l();
+            }
+
+            saveOutputTracks();
+
+            // put audio hardware into standby after short delay
+            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+                        isSuspended())) {
+                if (!mStandby) {
+
+                    threadLoop_standby();
+
+                    mStandby = true;
+                }
+
+                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+
+                    clearOutputTracks();
+
+                    if (exitPending()) {
+                        break;
+                    }
+
+                    releaseWakeLock_l();
+                    // wait until we have something to do...
+                    ALOGV("%s going to sleep", myName.string());
+                    mWaitWorkCV.wait(mLock);
+                    ALOGV("%s waking up", myName.string());
+                    acquireWakeLock_l();
+
+                    mMixerStatus = MIXER_IDLE;
+                    mMixerStatusIgnoringFastTracks = MIXER_IDLE;
+                    mBytesWritten = 0;
+
+                    checkSilentMode_l();
+
+                    standbyTime = systemTime() + standbyDelay;
+                    sleepTime = idleSleepTime;
+                    if (mType == MIXER) {
+                        sleepTimeShift = 0;
+                    }
+
+                    continue;
+                }
+            }
+
+            // mMixerStatusIgnoringFastTracks is also updated internally
+            mMixerStatus = prepareTracks_l(&tracksToRemove);
+
+            // prevent any changes in effect chain list and in each effect chain
+            // during mixing and effect process as the audio buffers could be deleted
+            // or modified if an effect is created or deleted
+            lockEffectChains_l(effectChains);
+        }
+
+        if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
+            threadLoop_mix();
+        } else {
+            threadLoop_sleepTime();
+        }
+
+        if (isSuspended()) {
+            sleepTime = suspendSleepTimeUs();
+            mBytesWritten += mixBufferSize;
+        }
+
+        // only process effects if we're going to write
+        if (sleepTime == 0) {
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+        }
+
+        // enable changes in effect chain
+        unlockEffectChains(effectChains);
+
+        // sleepTime == 0 means we must write to audio hardware
+        if (sleepTime == 0) {
+
+            threadLoop_write();
+
+if (mType == MIXER) {
+            // write blocked detection
+            nsecs_t now = systemTime();
+            nsecs_t delta = now - mLastWriteTime;
+            if (!mStandby && delta > maxPeriod) {
+                mNumDelayedWrites++;
+                if ((now - lastWarning) > kWarningThrottleNs) {
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+                    ScopedTrace st(ATRACE_TAG, "underrun");
+#endif
+                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+                            ns2ms(delta), mNumDelayedWrites, this);
+                    lastWarning = now;
+                }
+            }
+}
+
+            mStandby = false;
+        } else {
+            usleep(sleepTime);
+        }
+
+        // Finally let go of removed track(s), without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.  This will also mutate and push a new fast mixer state.
+        threadLoop_removeTracks(tracksToRemove);
+        tracksToRemove.clear();
+
+        // FIXME I don't understand the need for this here;
+        //       it was in the original code but maybe the
+        //       assignment in saveOutputTracks() makes this unnecessary?
+        clearOutputTracks();
+
+        // Effect chains will be actually deleted here if they were removed from
+        // mEffectChains list during mixing or effects processing
+        effectChains.clear();
+
+        // FIXME Note that the above .clear() is no longer necessary since effectChains
+        // is now local to this block, but will keep it for now (at least until merge done).
+    }
+
+    // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ...
+    if (mType == MIXER || mType == DIRECT) {
+        // put output stream into standby mode
+        if (!mStandby) {
+            mOutput->stream->common.standby(&mOutput->stream->common);
+        }
+    }
+
+    releaseWakeLock();
+
+    ALOGV("Thread %p type %d exiting", this, mType);
+    return false;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+        audio_io_handle_t id, audio_devices_t device, type_t type)
+    :   PlaybackThread(audioFlinger, output, id, device, type),
+        // mAudioMixer below
+        // mFastMixer below
+        mFastMixerFutex(0)
+        // mOutputSink below
+        // mPipeSink below
+        // mNormalSink below
+{
+    ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
+    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
+            "mFrameCount=%d, mNormalFrameCount=%d",
+            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
+            mNormalFrameCount);
+    mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+
+    // FIXME - Current mixer implementation only supports stereo output
+    if (mChannelCount != FCC_2) {
+        ALOGE("Invalid audio hardware channel count %d", mChannelCount);
+    }
+
+    // create an NBAIO sink for the HAL output stream, and negotiate
+    mOutputSink = new AudioStreamOutSink(output->stream);
+    size_t numCounterOffers = 0;
+    const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
+    ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
+    ALOG_ASSERT(index == 0);
+
+    // initialize fast mixer depending on configuration
+    bool initFastMixer;
+    switch (kUseFastMixer) {
+    case FastMixer_Never:
+        initFastMixer = false;
+        break;
+    case FastMixer_Always:
+        initFastMixer = true;
+        break;
+    case FastMixer_Static:
+    case FastMixer_Dynamic:
+        initFastMixer = mFrameCount < mNormalFrameCount;
+        break;
+    }
+    if (initFastMixer) {
+
+        // create a MonoPipe to connect our submix to FastMixer
+        NBAIO_Format format = mOutputSink->format();
+        // This pipe depth compensates for scheduling latency of the normal mixer thread.
+        // When it wakes up after a maximum latency, it runs a few cycles quickly before
+        // finally blocking.  Note the pipe implementation rounds up the request to a power of 2.
+        MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
+        const NBAIO_Format offers[1] = {format};
+        size_t numCounterOffers = 0;
+        ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        monoPipe->setAvgFrames((mScreenState & 1) ?
+                (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+        mPipeSink = monoPipe;
+
+#ifdef TEE_SINK_FRAMES
+        // create a Pipe to archive a copy of FastMixer's output for dumpsys
+        Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format);
+        numCounterOffers = 0;
+        index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        mTeeSink = teeSink;
+        PipeReader *teeSource = new PipeReader(*teeSink);
+        numCounterOffers = 0;
+        index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        mTeeSource = teeSource;
+#endif
+
+        // create fast mixer and configure it initially with just one fast track for our submix
+        mFastMixer = new FastMixer();
+        FastMixerStateQueue *sq = mFastMixer->sq();
+#ifdef STATE_QUEUE_DUMP
+        sq->setObserverDump(&mStateQueueObserverDump);
+        sq->setMutatorDump(&mStateQueueMutatorDump);
+#endif
+        FastMixerState *state = sq->begin();
+        FastTrack *fastTrack = &state->mFastTracks[0];
+        // wrap the source side of the MonoPipe to make it an AudioBufferProvider
+        fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
+        fastTrack->mVolumeProvider = NULL;
+        fastTrack->mGeneration++;
+        state->mFastTracksGen++;
+        state->mTrackMask = 1;
+        // fast mixer will use the HAL output sink
+        state->mOutputSink = mOutputSink.get();
+        state->mOutputSinkGen++;
+        state->mFrameCount = mFrameCount;
+        state->mCommand = FastMixerState::COLD_IDLE;
+        // already done in constructor initialization list
+        //mFastMixerFutex = 0;
+        state->mColdFutexAddr = &mFastMixerFutex;
+        state->mColdGen++;
+        state->mDumpState = &mFastMixerDumpState;
+        state->mTeeSink = mTeeSink.get();
+        sq->end();
+        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+
+        // start the fast mixer
+        mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
+        pid_t tid = mFastMixer->getTid();
+        int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+        if (err != 0) {
+            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+                    kPriorityFastMixer, getpid_cached, tid, err);
+        }
+
+#ifdef AUDIO_WATCHDOG
+        // create and start the watchdog
+        mAudioWatchdog = new AudioWatchdog();
+        mAudioWatchdog->setDump(&mAudioWatchdogDump);
+        mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
+        tid = mAudioWatchdog->getTid();
+        err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+        if (err != 0) {
+            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+                    kPriorityFastMixer, getpid_cached, tid, err);
+        }
+#endif
+
+    } else {
+        mFastMixer = NULL;
+    }
+
+    switch (kUseFastMixer) {
+    case FastMixer_Never:
+    case FastMixer_Dynamic:
+        mNormalSink = mOutputSink;
+        break;
+    case FastMixer_Always:
+        mNormalSink = mPipeSink;
+        break;
+    case FastMixer_Static:
+        mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
+        break;
+    }
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+    if (mFastMixer != NULL) {
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        if (state->mCommand == FastMixerState::COLD_IDLE) {
+            int32_t old = android_atomic_inc(&mFastMixerFutex);
+            if (old == -1) {
+                __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+            }
+        }
+        state->mCommand = FastMixerState::EXIT;
+        sq->end();
+        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+        mFastMixer->join();
+        // Though the fast mixer thread has exited, it's state queue is still valid.
+        // We'll use that extract the final state which contains one remaining fast track
+        // corresponding to our sub-mix.
+        state = sq->begin();
+        ALOG_ASSERT(state->mTrackMask == 1);
+        FastTrack *fastTrack = &state->mFastTracks[0];
+        ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
+        delete fastTrack->mBufferProvider;
+        sq->end(false /*didModify*/);
+        delete mFastMixer;
+#ifdef AUDIO_WATCHDOG
+        if (mAudioWatchdog != 0) {
+            mAudioWatchdog->requestExit();
+            mAudioWatchdog->requestExitAndWait();
+            mAudioWatchdog.clear();
+        }
+#endif
+    }
+    delete mAudioMixer;
+}
+
+
+uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
+{
+    if (mFastMixer != NULL) {
+        MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+        latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
+    }
+    return latency;
+}
+
+
+void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
+{
+    PlaybackThread::threadLoop_removeTracks(tracksToRemove);
+}
+
+void AudioFlinger::MixerThread::threadLoop_write()
+{
+    // FIXME we should only do one push per cycle; confirm this is true
+    // Start the fast mixer if it's not already running
+    if (mFastMixer != NULL) {
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        if (state->mCommand != FastMixerState::MIX_WRITE &&
+                (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
+            if (state->mCommand == FastMixerState::COLD_IDLE) {
+                int32_t old = android_atomic_inc(&mFastMixerFutex);
+                if (old == -1) {
+                    __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+                }
+#ifdef AUDIO_WATCHDOG
+                if (mAudioWatchdog != 0) {
+                    mAudioWatchdog->resume();
+                }
+#endif
+            }
+            state->mCommand = FastMixerState::MIX_WRITE;
+            sq->end();
+            sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+            if (kUseFastMixer == FastMixer_Dynamic) {
+                mNormalSink = mPipeSink;
+            }
+        } else {
+            sq->end(false /*didModify*/);
+        }
+    }
+    PlaybackThread::threadLoop_write();
+}
+
+void AudioFlinger::MixerThread::threadLoop_standby()
+{
+    // Idle the fast mixer if it's currently running
+    if (mFastMixer != NULL) {
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        if (!(state->mCommand & FastMixerState::IDLE)) {
+            state->mCommand = FastMixerState::COLD_IDLE;
+            state->mColdFutexAddr = &mFastMixerFutex;
+            state->mColdGen++;
+            mFastMixerFutex = 0;
+            sq->end();
+            // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
+            sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
+            if (kUseFastMixer == FastMixer_Dynamic) {
+                mNormalSink = mOutputSink;
+            }
+#ifdef AUDIO_WATCHDOG
+            if (mAudioWatchdog != 0) {
+                mAudioWatchdog->pause();
+            }
+#endif
+        } else {
+            sq->end(false /*didModify*/);
+        }
+    }
+    PlaybackThread::threadLoop_standby();
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_standby()
+{
+    ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
+    mOutput->stream->common.standby(&mOutput->stream->common);
+}
+
+void AudioFlinger::MixerThread::threadLoop_mix()
+{
+    // obtain the presentation timestamp of the next output buffer
+    int64_t pts;
+    status_t status = INVALID_OPERATION;
+
+    if (mNormalSink != 0) {
+        status = mNormalSink->getNextWriteTimestamp(&pts);
+    } else {
+        status = mOutputSink->getNextWriteTimestamp(&pts);
+    }
+
+    if (status != NO_ERROR) {
+        pts = AudioBufferProvider::kInvalidPTS;
+    }
+
+    // mix buffers...
+    mAudioMixer->process(pts);
+    // increase sleep time progressively when application underrun condition clears.
+    // Only increase sleep time if the mixer is ready for two consecutive times to avoid
+    // that a steady state of alternating ready/not ready conditions keeps the sleep time
+    // such that we would underrun the audio HAL.
+    if ((sleepTime == 0) && (sleepTimeShift > 0)) {
+        sleepTimeShift--;
+    }
+    sleepTime = 0;
+    standbyTime = systemTime() + standbyDelay;
+    //TODO: delay standby when effects have a tail
+}
+
+void AudioFlinger::MixerThread::threadLoop_sleepTime()
+{
+    // If no tracks are ready, sleep once for the duration of an output
+    // buffer size, then write 0s to the output
+    if (sleepTime == 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            sleepTime = activeSleepTime >> sleepTimeShift;
+            if (sleepTime < kMinThreadSleepTimeUs) {
+                sleepTime = kMinThreadSleepTimeUs;
+            }
+            // reduce sleep time in case of consecutive application underruns to avoid
+            // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
+            // duration we would end up writing less data than needed by the audio HAL if
+            // the condition persists.
+            if (sleepTimeShift < kMaxThreadSleepTimeShift) {
+                sleepTimeShift++;
+            }
+        } else {
+            sleepTime = idleSleepTime;
+        }
+    } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
+        memset (mMixBuffer, 0, mixBufferSize);
+        sleepTime = 0;
+        ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
+                "anticipated start");
+    }
+    // TODO add standby time extension fct of effect tail
+}
+
+// prepareTracks_l() must be called with ThreadBase::mLock held
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+        Vector< sp<Track> > *tracksToRemove)
+{
+
+    mixer_state mixerStatus = MIXER_IDLE;
+    // find out which tracks need to be processed
+    size_t count = mActiveTracks.size();
+    size_t mixedTracks = 0;
+    size_t tracksWithEffect = 0;
+    // counts only _active_ fast tracks
+    size_t fastTracks = 0;
+    uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
+
+    float masterVolume = mMasterVolume;
+    bool masterMute = mMasterMute;
+
+    if (masterMute) {
+        masterVolume = 0;
+    }
+    // Delegate master volume control to effect in output mix effect chain if needed
+    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
+    if (chain != 0) {
+        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
+        chain->setVolume_l(&v, &v);
+        masterVolume = (float)((v + (1 << 23)) >> 24);
+        chain.clear();
+    }
+
+    // prepare a new state to push
+    FastMixerStateQueue *sq = NULL;
+    FastMixerState *state = NULL;
+    bool didModify = false;
+    FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
+    if (mFastMixer != NULL) {
+        sq = mFastMixer->sq();
+        state = sq->begin();
+    }
+
+    for (size_t i=0 ; i<count ; i++) {
+        sp<Track> t = mActiveTracks[i].promote();
+        if (t == 0) {
+            continue;
+        }
+
+        // this const just means the local variable doesn't change
+        Track* const track = t.get();
+
+        // process fast tracks
+        if (track->isFastTrack()) {
+
+            // It's theoretically possible (though unlikely) for a fast track to be created
+            // and then removed within the same normal mix cycle.  This is not a problem, as
+            // the track never becomes active so it's fast mixer slot is never touched.
+            // The converse, of removing an (active) track and then creating a new track
+            // at the identical fast mixer slot within the same normal mix cycle,
+            // is impossible because the slot isn't marked available until the end of each cycle.
+            int j = track->mFastIndex;
+            ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks);
+            ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
+            FastTrack *fastTrack = &state->mFastTracks[j];
+
+            // Determine whether the track is currently in underrun condition,
+            // and whether it had a recent underrun.
+            FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
+            FastTrackUnderruns underruns = ftDump->mUnderruns;
+            uint32_t recentFull = (underruns.mBitFields.mFull -
+                    track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
+            uint32_t recentPartial = (underruns.mBitFields.mPartial -
+                    track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
+            uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
+                    track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
+            uint32_t recentUnderruns = recentPartial + recentEmpty;
+            track->mObservedUnderruns = underruns;
+            // don't count underruns that occur while stopping or pausing
+            // or stopped which can occur when flush() is called while active
+            if (!(track->isStopping() || track->isPausing() || track->isStopped())) {
+                track->mUnderrunCount += recentUnderruns;
+            }
+
+            // This is similar to the state machine for normal tracks,
+            // with a few modifications for fast tracks.
+            bool isActive = true;
+            switch (track->mState) {
+            case TrackBase::STOPPING_1:
+                // track stays active in STOPPING_1 state until first underrun
+                if (recentUnderruns > 0) {
+                    track->mState = TrackBase::STOPPING_2;
+                }
+                break;
+            case TrackBase::PAUSING:
+                // ramp down is not yet implemented
+                track->setPaused();
+                break;
+            case TrackBase::RESUMING:
+                // ramp up is not yet implemented
+                track->mState = TrackBase::ACTIVE;
+                break;
+            case TrackBase::ACTIVE:
+                if (recentFull > 0 || recentPartial > 0) {
+                    // track has provided at least some frames recently: reset retry count
+                    track->mRetryCount = kMaxTrackRetries;
+                }
+                if (recentUnderruns == 0) {
+                    // no recent underruns: stay active
+                    break;
+                }
+                // there has recently been an underrun of some kind
+                if (track->sharedBuffer() == 0) {
+                    // were any of the recent underruns "empty" (no frames available)?
+                    if (recentEmpty == 0) {
+                        // no, then ignore the partial underruns as they are allowed indefinitely
+                        break;
+                    }
+                    // there has recently been an "empty" underrun: decrement the retry counter
+                    if (--(track->mRetryCount) > 0) {
+                        break;
+                    }
+                    // indicate to client process that the track was disabled because of underrun;
+                    // it will then automatically call start() when data is available
+                    android_atomic_or(CBLK_DISABLED, &track->mCblk->flags);
+                    // remove from active list, but state remains ACTIVE [confusing but true]
+                    isActive = false;
+                    break;
+                }
+                // fall through
+            case TrackBase::STOPPING_2:
+            case TrackBase::PAUSED:
+            case TrackBase::TERMINATED:
+            case TrackBase::STOPPED:
+            case TrackBase::FLUSHED:   // flush() while active
+                // Check for presentation complete if track is inactive
+                // We have consumed all the buffers of this track.
+                // This would be incomplete if we auto-paused on underrun
+                {
+                    size_t audioHALFrames =
+                            (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
+                    size_t framesWritten = mBytesWritten / mFrameSize;
+                    if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
+                        // track stays in active list until presentation is complete
+                        break;
+                    }
+                }
+                if (track->isStopping_2()) {
+                    track->mState = TrackBase::STOPPED;
+                }
+                if (track->isStopped()) {
+                    // Can't reset directly, as fast mixer is still polling this track
+                    //   track->reset();
+                    // So instead mark this track as needing to be reset after push with ack
+                    resetMask |= 1 << i;
+                }
+                isActive = false;
+                break;
+            case TrackBase::IDLE:
+            default:
+                LOG_FATAL("unexpected track state %d", track->mState);
+            }
+
+            if (isActive) {
+                // was it previously inactive?
+                if (!(state->mTrackMask & (1 << j))) {
+                    ExtendedAudioBufferProvider *eabp = track;
+                    VolumeProvider *vp = track;
+                    fastTrack->mBufferProvider = eabp;
+                    fastTrack->mVolumeProvider = vp;
+                    fastTrack->mSampleRate = track->mSampleRate;
+                    fastTrack->mChannelMask = track->mChannelMask;
+                    fastTrack->mGeneration++;
+                    state->mTrackMask |= 1 << j;
+                    didModify = true;
+                    // no acknowledgement required for newly active tracks
+                }
+                // cache the combined master volume and stream type volume for fast mixer; this
+                // lacks any synchronization or barrier so VolumeProvider may read a stale value
+                track->mCachedVolume = track->isMuted() ?
+                        0 : masterVolume * mStreamTypes[track->streamType()].volume;
+                ++fastTracks;
+            } else {
+                // was it previously active?
+                if (state->mTrackMask & (1 << j)) {
+                    fastTrack->mBufferProvider = NULL;
+                    fastTrack->mGeneration++;
+                    state->mTrackMask &= ~(1 << j);
+                    didModify = true;
+                    // If any fast tracks were removed, we must wait for acknowledgement
+                    // because we're about to decrement the last sp<> on those tracks.
+                    block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
+                } else {
+                    LOG_FATAL("fast track %d should have been active", j);
+                }
+                tracksToRemove->add(track);
+                // Avoids a misleading display in dumpsys
+                track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
+            }
+            continue;
+        }
+
+        {   // local variable scope to avoid goto warning
+
+        audio_track_cblk_t* cblk = track->cblk();
+
+        // The first time a track is added we wait
+        // for all its buffers to be filled before processing it
+        int name = track->name();
+        // make sure that we have enough frames to mix one full buffer.
+        // enforce this condition only once to enable draining the buffer in case the client
+        // app does not call stop() and relies on underrun to stop:
+        // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
+        // during last round
+        uint32_t minFrames = 1;
+        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
+                (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
+            if (t->sampleRate() == mSampleRate) {
+                minFrames = mNormalFrameCount;
+            } else {
+                // +1 for rounding and +1 for additional sample needed for interpolation
+                minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+                // add frames already consumed but not yet released by the resampler
+                // because cblk->framesReady() will include these frames
+                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
+                // the minimum track buffer size is normally twice the number of frames necessary
+                // to fill one buffer and the resampler should not leave more than one buffer worth
+                // of unreleased frames after each pass, but just in case...
+                ALOG_ASSERT(minFrames <= cblk->frameCount);
+            }
+        }
+        if ((track->framesReady() >= minFrames) && track->isReady() &&
+                !track->isPaused() && !track->isTerminated())
+        {
+            ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
+                    this);
+
+            mixedTracks++;
+
+            // track->mainBuffer() != mMixBuffer means there is an effect chain
+            // connected to the track
+            chain.clear();
+            if (track->mainBuffer() != mMixBuffer) {
+                chain = getEffectChain_l(track->sessionId());
+                // Delegate volume control to effect in track effect chain if needed
+                if (chain != 0) {
+                    tracksWithEffect++;
+                } else {
+                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
+                            "session %d",
+                            name, track->sessionId());
+                }
+            }
+
+
+            int param = AudioMixer::VOLUME;
+            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;
+                }
+                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
+            } else if (cblk->server != 0) {
+                // If the track is stopped before the first frame was mixed,
+                // do not apply ramp
+                param = AudioMixer::RAMP_VOLUME;
+            }
+
+            // compute volume for this track
+            uint32_t vl, vr, va;
+            if (track->isMuted() || track->isPausing() ||
+                mStreamTypes[track->streamType()].mute) {
+                vl = vr = va = 0;
+                if (track->isPausing()) {
+                    track->setPaused();
+                }
+            } else {
+
+                // read original volumes with volume control
+                float typeVolume = mStreamTypes[track->streamType()].volume;
+                float v = masterVolume * typeVolume;
+                uint32_t vlr = cblk->getVolumeLR();
+                vl = vlr & 0xFFFF;
+                vr = vlr >> 16;
+                // track volumes come from shared memory, so can't be trusted and must be clamped
+                if (vl > MAX_GAIN_INT) {
+                    ALOGV("Track left volume out of range: %04X", vl);
+                    vl = MAX_GAIN_INT;
+                }
+                if (vr > MAX_GAIN_INT) {
+                    ALOGV("Track right volume out of range: %04X", vr);
+                    vr = MAX_GAIN_INT;
+                }
+                // now apply the master volume and stream type volume
+                vl = (uint32_t)(v * vl) << 12;
+                vr = (uint32_t)(v * vr) << 12;
+                // assuming master volume and stream type volume each go up to 1.0,
+                // vl and vr are now in 8.24 format
+
+                uint16_t sendLevel = cblk->getSendLevel_U4_12();
+                // send level comes from shared memory and so may be corrupt
+                if (sendLevel > MAX_GAIN_INT) {
+                    ALOGV("Track send level out of range: %04X", sendLevel);
+                    sendLevel = MAX_GAIN_INT;
+                }
+                va = (uint32_t)(v * sendLevel);
+            }
+            // Delegate volume control to effect in track effect chain if needed
+            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
+                // Do not ramp volume if volume is controlled by effect
+                param = AudioMixer::VOLUME;
+                track->mHasVolumeController = true;
+            } else {
+                // force no volume ramp when volume controller was just disabled or removed
+                // from effect chain to avoid volume spike
+                if (track->mHasVolumeController) {
+                    param = AudioMixer::VOLUME;
+                }
+                track->mHasVolumeController = false;
+            }
+
+            // Convert volumes from 8.24 to 4.12 format
+            // This additional clamping is needed in case chain->setVolume_l() overshot
+            vl = (vl + (1 << 11)) >> 12;
+            if (vl > MAX_GAIN_INT) {
+                vl = MAX_GAIN_INT;
+            }
+            vr = (vr + (1 << 11)) >> 12;
+            if (vr > MAX_GAIN_INT) {
+                vr = MAX_GAIN_INT;
+            }
+
+            if (va > MAX_GAIN_INT) {
+                va = MAX_GAIN_INT;   // va is uint32_t, so no need to check for -
+            }
+
+            // XXX: these things DON'T need to be done each time
+            mAudioMixer->setBufferProvider(name, track);
+            mAudioMixer->enable(name);
+
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
+            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::FORMAT, (void *)track->format());
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::RESAMPLE,
+                AudioMixer::SAMPLE_RATE,
+                (void *)(cblk->sampleRate));
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
+
+            // reset retry count
+            track->mRetryCount = kMaxTrackRetries;
+
+            // If one track is ready, set the mixer ready if:
+            //  - the mixer was not ready during previous round OR
+            //  - no other track is not ready
+            if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
+                    mixerStatus != MIXER_TRACKS_ENABLED) {
+                mixerStatus = MIXER_TRACKS_READY;
+            }
+        } else {
+            // clear effect chain input buffer if an active track underruns to avoid sending
+            // previous audio buffer again to effects
+            chain = getEffectChain_l(track->sessionId());
+            if (chain != 0) {
+                chain->clearInputBuffer();
+            }
+
+            ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user,
+                    cblk->server, this);
+            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
+                    track->isStopped() || track->isPaused()) {
+                // We have consumed all the buffers of this track.
+                // Remove it from the list of active tracks.
+                // TODO: use actual buffer filling status instead of latency when available from
+                // audio HAL
+                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+                size_t framesWritten = mBytesWritten / mFrameSize;
+                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+                    if (track->isStopped()) {
+                        track->reset();
+                    }
+                    tracksToRemove->add(track);
+                }
+            } else {
+                track->mUnderrunCount++;
+                // No buffers for this track. Give it a few chances to
+                // fill a buffer, then remove it from active list.
+                if (--(track->mRetryCount) <= 0) {
+                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
+                    tracksToRemove->add(track);
+                    // indicate to client process that the track was disabled because of underrun;
+                    // it will then automatically call start() when data is available
+                    android_atomic_or(CBLK_DISABLED, &cblk->flags);
+                // If one track is not ready, mark the mixer also not ready if:
+                //  - the mixer was ready during previous round OR
+                //  - no other track is ready
+                } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
+                                mixerStatus != MIXER_TRACKS_READY) {
+                    mixerStatus = MIXER_TRACKS_ENABLED;
+                }
+            }
+            mAudioMixer->disable(name);
+        }
+
+        }   // local variable scope to avoid goto warning
+track_is_ready: ;
+
+    }
+
+    // Push the new FastMixer state if necessary
+    bool pauseAudioWatchdog = false;
+    if (didModify) {
+        state->mFastTracksGen++;
+        // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
+        if (kUseFastMixer == FastMixer_Dynamic &&
+                state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
+            state->mCommand = FastMixerState::COLD_IDLE;
+            state->mColdFutexAddr = &mFastMixerFutex;
+            state->mColdGen++;
+            mFastMixerFutex = 0;
+            if (kUseFastMixer == FastMixer_Dynamic) {
+                mNormalSink = mOutputSink;
+            }
+            // If we go into cold idle, need to wait for acknowledgement
+            // so that fast mixer stops doing I/O.
+            block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
+            pauseAudioWatchdog = true;
+        }
+        sq->end();
+    }
+    if (sq != NULL) {
+        sq->end(didModify);
+        sq->push(block);
+    }
+#ifdef AUDIO_WATCHDOG
+    if (pauseAudioWatchdog && mAudioWatchdog != 0) {
+        mAudioWatchdog->pause();
+    }
+#endif
+
+    // Now perform the deferred reset on fast tracks that have stopped
+    while (resetMask != 0) {
+        size_t i = __builtin_ctz(resetMask);
+        ALOG_ASSERT(i < count);
+        resetMask &= ~(1 << i);
+        sp<Track> t = mActiveTracks[i].promote();
+        if (t == 0) {
+            continue;
+        }
+        Track* track = t.get();
+        ALOG_ASSERT(track->isFastTrack() && track->isStopped());
+        track->reset();
+    }
+
+    // remove all the tracks that need to be...
+    count = tracksToRemove->size();
+    if (CC_UNLIKELY(count)) {
+        for (size_t i=0 ; i<count ; i++) {
+            const sp<Track>& track = tracksToRemove->itemAt(i);
+            mActiveTracks.remove(track);
+            if (track->mainBuffer() != mMixBuffer) {
+                chain = getEffectChain_l(track->sessionId());
+                if (chain != 0) {
+                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
+                            track->sessionId());
+                    chain->decActiveTrackCnt();
+                }
+            }
+            if (track->isTerminated()) {
+                removeTrack_l(track);
+            }
+        }
+    }
+
+    // mix buffer must be cleared if all tracks are connected to an
+    // effect chain as in this case the mixer will not write to
+    // mix buffer and track effects will accumulate into it
+    if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
+            (mixedTracks == 0 && fastTracks > 0)) {
+        // FIXME as a performance optimization, should remember previous zero status
+        memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+    }
+
+    // if any fast tracks, then status is ready
+    mMixerStatusIgnoringFastTracks = mixerStatus;
+    if (fastTracks > 0) {
+        mixerStatus = MIXER_TRACKS_READY;
+    }
+    return mixerStatus;
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
+{
+    return mAudioMixer->getTrackName(channelMask, sessionId);
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::MixerThread::deleteTrackName_l(int name)
+{
+    ALOGV("remove track (%d) and delete from mixer", name);
+    mAudioMixer->deleteTrackName(name);
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
+{
+    // if !&IDLE, holds the FastMixer state to restore after new parameters processed
+    FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+
+        if (mFastMixer != NULL) {
+            FastMixerStateQueue *sq = mFastMixer->sq();
+            FastMixerState *state = sq->begin();
+            if (!(state->mCommand & FastMixerState::IDLE)) {
+                previousCommand = state->mCommand;
+                state->mCommand = FastMixerState::HOT_IDLE;
+                sq->end();
+                sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
+            } else {
+                sq->end(false /*didModify*/);
+            }
+        }
+
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            if (value != AUDIO_CHANNEL_OUT_STEREO) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be guaranteed
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+#ifdef ADD_BATTERY_DATA
+            // when changing the audio output device, call addBatteryData to notify
+            // the change
+            if (mOutDevice != value) {
+                uint32_t params = 0;
+                // check whether speaker is on
+                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
+                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
+                }
+
+                audio_devices_t deviceWithoutSpeaker
+                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
+                // check if any other device (except speaker) is on
+                if (value & deviceWithoutSpeaker ) {
+                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
+                }
+
+                if (params != 0) {
+                    addBatteryData(params);
+                }
+            }
+#endif
+
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            mOutDevice = value;
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setDevice_l(mOutDevice);
+            }
+        }
+
+        if (status == NO_ERROR) {
+            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                    keyValuePair.string());
+            if (!mStandby && status == INVALID_OPERATION) {
+                mOutput->stream->common.standby(&mOutput->stream->common);
+                mStandby = true;
+                mBytesWritten = 0;
+                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                       keyValuePair.string());
+            }
+            if (status == NO_ERROR && reconfig) {
+                delete mAudioMixer;
+                // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+                mAudioMixer = NULL;
+                readOutputParameters();
+                mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+                for (size_t i = 0; i < mTracks.size() ; i++) {
+                    int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
+                    if (name < 0) {
+                        break;
+                    }
+                    mTracks[i]->mName = name;
+                    // limit track sample rate to 2 x new output sample rate
+                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
+                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
+                    }
+                }
+                sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+        // already timed out waiting for the status and will never signal the condition.
+        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+    }
+
+    if (!(previousCommand & FastMixerState::IDLE)) {
+        ALOG_ASSERT(mFastMixer != NULL);
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
+        state->mCommand = previousCommand;
+        sq->end();
+        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+    }
+
+    return reconfig;
+}
+
+
+void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    PlaybackThread::dumpInternals(fd, args);
+
+    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
+    FastMixerDumpState copy = mFastMixerDumpState;
+    copy.dump(fd);
+
+#ifdef STATE_QUEUE_DUMP
+    // Similar for state queue
+    StateQueueObserverDump observerCopy = mStateQueueObserverDump;
+    observerCopy.dump(fd);
+    StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
+    mutatorCopy.dump(fd);
+#endif
+
+    // Write the tee output to a .wav file
+    dumpTee(fd, mTeeSource, mId);
+
+#ifdef AUDIO_WATCHDOG
+    if (mAudioWatchdog != 0) {
+        // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
+        AudioWatchdogDump wdCopy = mAudioWatchdogDump;
+        wdCopy.dump(fd);
+    }
+#endif
+}
+
+uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
+{
+    return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
+}
+
+uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
+{
+    return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
+}
+
+void AudioFlinger::MixerThread::cacheParameters_l()
+{
+    PlaybackThread::cacheParameters_l();
+
+    // FIXME: Relaxed timing because of a certain device that can't meet latency
+    // Should be reduced to 2x after the vendor fixes the driver issue
+    // increase threshold again due to low power audio mode. The way this warning
+    // threshold is calculated and its usefulness should be reconsidered anyway.
+    maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+        AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
+    :   PlaybackThread(audioFlinger, output, id, device, DIRECT)
+        // mLeftVolFloat, mRightVolFloat
+{
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
+    Vector< sp<Track> > *tracksToRemove
+)
+{
+    sp<Track> trackToRemove;
+
+    mixer_state mixerStatus = MIXER_IDLE;
+
+    // find out which tracks need to be processed
+    if (mActiveTracks.size() != 0) {
+        sp<Track> t = mActiveTracks[0].promote();
+        // The track died recently
+        if (t == 0) {
+            return MIXER_IDLE;
+        }
+
+        Track* const track = t.get();
+        audio_track_cblk_t* cblk = track->cblk();
+
+        // The first time a track is added we wait
+        // for all its buffers to be filled before processing it
+        uint32_t minFrames;
+        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
+            minFrames = mNormalFrameCount;
+        } else {
+            minFrames = 1;
+        }
+        if ((track->framesReady() >= minFrames) && track->isReady() &&
+                !track->isPaused() && !track->isTerminated())
+        {
+            ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+            if (track->mFillingUpStatus == Track::FS_FILLED) {
+                track->mFillingUpStatus = Track::FS_ACTIVE;
+                mLeftVolFloat = mRightVolFloat = 0;
+                if (track->mState == TrackBase::RESUMING) {
+                    track->mState = TrackBase::ACTIVE;
+                }
+            }
+
+            // compute volume for this track
+            float left, right;
+            if (track->isMuted() || mMasterMute || track->isPausing() ||
+                mStreamTypes[track->streamType()].mute) {
+                left = right = 0;
+                if (track->isPausing()) {
+                    track->setPaused();
+                }
+            } else {
+                float typeVolume = mStreamTypes[track->streamType()].volume;
+                float v = mMasterVolume * typeVolume;
+                uint32_t vlr = cblk->getVolumeLR();
+                float v_clamped = v * (vlr & 0xFFFF);
+                if (v_clamped > MAX_GAIN) {
+                    v_clamped = MAX_GAIN;
+                }
+                left = v_clamped/MAX_GAIN;
+                v_clamped = v * (vlr >> 16);
+                if (v_clamped > MAX_GAIN) {
+                    v_clamped = MAX_GAIN;
+                }
+                right = v_clamped/MAX_GAIN;
+            }
+
+            if (left != mLeftVolFloat || right != mRightVolFloat) {
+                mLeftVolFloat = left;
+                mRightVolFloat = right;
+
+                // Convert volumes from float to 8.24
+                uint32_t vl = (uint32_t)(left * (1 << 24));
+                uint32_t vr = (uint32_t)(right * (1 << 24));
+
+                // Delegate volume control to effect in track effect chain if needed
+                // only one effect chain can be present on DirectOutputThread, so if
+                // there is one, the track is connected to it
+                if (!mEffectChains.isEmpty()) {
+                    // Do not ramp volume if volume is controlled by effect
+                    mEffectChains[0]->setVolume_l(&vl, &vr);
+                    left = (float)vl / (1 << 24);
+                    right = (float)vr / (1 << 24);
+                }
+                mOutput->stream->set_volume(mOutput->stream, left, right);
+            }
+
+            // reset retry count
+            track->mRetryCount = kMaxTrackRetriesDirect;
+            mActiveTrack = t;
+            mixerStatus = MIXER_TRACKS_READY;
+        } else {
+            // clear effect chain input buffer if an active track underruns to avoid sending
+            // previous audio buffer again to effects
+            if (!mEffectChains.isEmpty()) {
+                mEffectChains[0]->clearInputBuffer();
+            }
+
+            ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
+                    track->isStopped() || track->isPaused()) {
+                // We have consumed all the buffers of this track.
+                // Remove it from the list of active tracks.
+                // TODO: implement behavior for compressed audio
+                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+                size_t framesWritten = mBytesWritten / mFrameSize;
+                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+                    if (track->isStopped()) {
+                        track->reset();
+                    }
+                    trackToRemove = 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) {
+                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+                    trackToRemove = track;
+                } else {
+                    mixerStatus = MIXER_TRACKS_ENABLED;
+                }
+            }
+        }
+    }
+
+    // FIXME merge this with similar code for removing multiple tracks
+    // remove all the tracks that need to be...
+    if (CC_UNLIKELY(trackToRemove != 0)) {
+        tracksToRemove->add(trackToRemove);
+        mActiveTracks.remove(trackToRemove);
+        if (!mEffectChains.isEmpty()) {
+            ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(),
+                    trackToRemove->sessionId());
+            mEffectChains[0]->decActiveTrackCnt();
+        }
+        if (trackToRemove->isTerminated()) {
+            removeTrack_l(trackToRemove);
+        }
+    }
+
+    return mixerStatus;
+}
+
+void AudioFlinger::DirectOutputThread::threadLoop_mix()
+{
+    AudioBufferProvider::Buffer buffer;
+    size_t frameCount = mFrameCount;
+    int8_t *curBuf = (int8_t *)mMixBuffer;
+    // output audio to hardware
+    while (frameCount) {
+        buffer.frameCount = frameCount;
+        mActiveTrack->getNextBuffer(&buffer);
+        if (CC_UNLIKELY(buffer.raw == NULL)) {
+            memset(curBuf, 0, frameCount * mFrameSize);
+            break;
+        }
+        memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+        frameCount -= buffer.frameCount;
+        curBuf += buffer.frameCount * mFrameSize;
+        mActiveTrack->releaseBuffer(&buffer);
+    }
+    sleepTime = 0;
+    standbyTime = systemTime() + standbyDelay;
+    mActiveTrack.clear();
+
+}
+
+void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
+{
+    if (sleepTime == 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            sleepTime = activeSleepTime;
+        } else {
+            sleepTime = idleSleepTime;
+        }
+    } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
+        memset(mMixBuffer, 0, mFrameCount * mFrameSize);
+        sleepTime = 0;
+    }
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
+        int sessionId)
+{
+    return 0;
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+{
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                    keyValuePair.string());
+            if (!mStandby && status == INVALID_OPERATION) {
+                mOutput->stream->common.standby(&mOutput->stream->common);
+                mStandby = true;
+                mBytesWritten = 0;
+                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                       keyValuePair.string());
+            }
+            if (status == NO_ERROR && reconfig) {
+                readOutputParameters();
+                sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+        // already timed out waiting for the status and will never signal the condition.
+        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+    }
+    return reconfig;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
+{
+    uint32_t time;
+    if (audio_is_linear_pcm(mFormat)) {
+        time = PlaybackThread::activeSleepTimeUs();
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
+{
+    uint32_t time;
+    if (audio_is_linear_pcm(mFormat)) {
+        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
+{
+    uint32_t time;
+    if (audio_is_linear_pcm(mFormat)) {
+        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+void AudioFlinger::DirectOutputThread::cacheParameters_l()
+{
+    PlaybackThread::cacheParameters_l();
+
+    // use shorter standby delay as on normal output to release
+    // hardware resources as soon as possible
+    standbyDelay = microseconds(activeSleepTime*2);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+        AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
+    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
+                DUPLICATING),
+        mWaitTimeMs(UINT_MAX)
+{
+    addOutputTrack(mainThread);
+}
+
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
+{
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        mOutputTracks[i]->destroy();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_mix()
+{
+    // mix buffers...
+    if (outputsReady(outputTracks)) {
+        mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
+    } else {
+        memset(mMixBuffer, 0, mixBufferSize);
+    }
+    sleepTime = 0;
+    writeFrames = mNormalFrameCount;
+    standbyTime = systemTime() + standbyDelay;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
+{
+    if (sleepTime == 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            sleepTime = activeSleepTime;
+        } else {
+            sleepTime = idleSleepTime;
+        }
+    } else if (mBytesWritten != 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            writeFrames = mNormalFrameCount;
+            memset(mMixBuffer, 0, mixBufferSize);
+        } else {
+            // flush remaining overflow buffers in output tracks
+            writeFrames = 0;
+        }
+        sleepTime = 0;
+    }
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_write()
+{
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        outputTracks[i]->write(mMixBuffer, writeFrames);
+    }
+    mBytesWritten += mixBufferSize;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_standby()
+{
+    // DuplicatingThread implements standby by stopping all tracks
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        outputTracks[i]->stop();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::saveOutputTracks()
+{
+    outputTracks = mOutputTracks;
+}
+
+void AudioFlinger::DuplicatingThread::clearOutputTracks()
+{
+    outputTracks.clear();
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+    Mutex::Autolock _l(mLock);
+    // FIXME explain this formula
+    size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
+    OutputTrack *outputTrack = new OutputTrack(thread,
+                                            this,
+                                            mSampleRate,
+                                            mFormat,
+                                            mChannelMask,
+                                            frameCount);
+    if (outputTrack->cblk() != NULL) {
+        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
+        mOutputTracks.add(outputTrack);
+        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+        updateWaitTime_l();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+{
+    Mutex::Autolock _l(mLock);
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        if (mOutputTracks[i]->thread() == thread) {
+            mOutputTracks[i]->destroy();
+            mOutputTracks.removeAt(i);
+            updateWaitTime_l();
+            return;
+        }
+    }
+    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
+}
+
+// caller must hold mLock
+void AudioFlinger::DuplicatingThread::updateWaitTime_l()
+{
+    mWaitTimeMs = UINT_MAX;
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
+        if (strong != 0) {
+            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
+            if (waitTimeMs < mWaitTimeMs) {
+                mWaitTimeMs = waitTimeMs;
+            }
+        }
+    }
+}
+
+
+bool AudioFlinger::DuplicatingThread::outputsReady(
+        const SortedVector< sp<OutputTrack> > &outputTracks)
+{
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        sp<ThreadBase> thread = outputTracks[i]->thread().promote();
+        if (thread == 0) {
+            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
+                    outputTracks[i].get());
+            return false;
+        }
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        // see note at standby() declaration
+        if (playbackThread->standby() && !playbackThread->isSuspended()) {
+            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
+                    thread.get());
+            return false;
+        }
+    }
+    return true;
+}
+
+uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
+{
+    return (mWaitTimeMs * 1000) / 2;
+}
+
+void AudioFlinger::DuplicatingThread::cacheParameters_l()
+{
+    // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
+    updateWaitTime_l();
+
+    MixerThread::cacheParameters_l();
+}
+
+// ----------------------------------------------------------------------------
+//      Record
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
+                                         AudioStreamIn *input,
+                                         uint32_t sampleRate,
+                                         audio_channel_mask_t channelMask,
+                                         audio_io_handle_t id,
+                                         audio_devices_t device,
+                                         const sp<NBAIO_Sink>& teeSink) :
+    ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD),
+    mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+    // mRsmpInIndex and mInputBytes set by readInputParameters()
+    mReqChannelCount(popcount(channelMask)),
+    mReqSampleRate(sampleRate),
+    // mBytesRead is only meaningful while active, and so is cleared in start()
+    // (but might be better to also clear here for dump?)
+    mTeeSink(teeSink)
+{
+    snprintf(mName, kNameLength, "AudioIn_%X", id);
+
+    readInputParameters();
+
+}
+
+
+AudioFlinger::RecordThread::~RecordThread()
+{
+    delete[] mRsmpInBuffer;
+    delete mResampler;
+    delete[] mRsmpOutBuffer;
+}
+
+void AudioFlinger::RecordThread::onFirstRef()
+{
+    run(mName, PRIORITY_URGENT_AUDIO);
+}
+
+status_t AudioFlinger::RecordThread::readyToRun()
+{
+    status_t status = initCheck();
+    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
+    return status;
+}
+
+bool AudioFlinger::RecordThread::threadLoop()
+{
+    AudioBufferProvider::Buffer buffer;
+    sp<RecordTrack> activeTrack;
+    Vector< sp<EffectChain> > effectChains;
+
+    nsecs_t lastWarning = 0;
+
+    inputStandBy();
+    acquireWakeLock();
+
+    // used to verify we've read at least once before evaluating how many bytes were read
+    bool readOnce = false;
+
+    // start recording
+    while (!exitPending()) {
+
+        processConfigEvents();
+
+        { // scope for mLock
+            Mutex::Autolock _l(mLock);
+            checkForNewParameters_l();
+            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+                standby();
+
+                if (exitPending()) {
+                    break;
+                }
+
+                releaseWakeLock_l();
+                ALOGV("RecordThread: loop stopping");
+                // go to sleep
+                mWaitWorkCV.wait(mLock);
+                ALOGV("RecordThread: loop starting");
+                acquireWakeLock_l();
+                continue;
+            }
+            if (mActiveTrack != 0) {
+                if (mActiveTrack->mState == TrackBase::PAUSING) {
+                    standby();
+                    mActiveTrack.clear();
+                    mStartStopCond.broadcast();
+                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+                    if (mReqChannelCount != mActiveTrack->channelCount()) {
+                        mActiveTrack.clear();
+                        mStartStopCond.broadcast();
+                    } else if (readOnce) {
+                        // record start succeeds only if first read from audio input
+                        // succeeds
+                        if (mBytesRead >= 0) {
+                            mActiveTrack->mState = TrackBase::ACTIVE;
+                        } else {
+                            mActiveTrack.clear();
+                        }
+                        mStartStopCond.broadcast();
+                    }
+                    mStandby = false;
+                } else if (mActiveTrack->mState == TrackBase::TERMINATED) {
+                    removeTrack_l(mActiveTrack);
+                    mActiveTrack.clear();
+                }
+            }
+            lockEffectChains_l(effectChains);
+        }
+
+        if (mActiveTrack != 0) {
+            if (mActiveTrack->mState != TrackBase::ACTIVE &&
+                mActiveTrack->mState != TrackBase::RESUMING) {
+                unlockEffectChains(effectChains);
+                usleep(kRecordThreadSleepUs);
+                continue;
+            }
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+
+            buffer.frameCount = mFrameCount;
+            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+                readOnce = true;
+                size_t framesOut = buffer.frameCount;
+                if (mResampler == NULL) {
+                    // no resampling
+                    while (framesOut) {
+                        size_t framesIn = mFrameCount - mRsmpInIndex;
+                        if (framesIn) {
+                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
+                                    mActiveTrack->mFrameSize;
+                            if (framesIn > framesOut)
+                                framesIn = framesOut;
+                            mRsmpInIndex += framesIn;
+                            framesOut -= framesIn;
+                            if (mChannelCount == mReqChannelCount ||
+                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+                                memcpy(dst, src, framesIn * mFrameSize);
+                            } else {
+                                if (mChannelCount == 1) {
+                                    upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
+                                            (int16_t *)src, framesIn);
+                                } else {
+                                    downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+                                            (int16_t *)src, framesIn);
+                                }
+                            }
+                        }
+                        if (framesOut && mFrameCount == mRsmpInIndex) {
+                            void *readInto;
+                            if (framesOut == mFrameCount &&
+                                (mChannelCount == mReqChannelCount ||
+                                        mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+                                readInto = buffer.raw;
+                                framesOut = 0;
+                            } else {
+                                readInto = mRsmpInBuffer;
+                                mRsmpInIndex = 0;
+                            }
+                            mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes);
+                            if (mBytesRead <= 0) {
+                                if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
+                                {
+                                    ALOGE("Error reading audio input");
+                                    // Force input into standby so that it tries to
+                                    // recover at next read attempt
+                                    inputStandBy();
+                                    usleep(kRecordThreadSleepUs);
+                                }
+                                mRsmpInIndex = mFrameCount;
+                                framesOut = 0;
+                                buffer.frameCount = 0;
+                            } else if (mTeeSink != 0) {
+                                (void) mTeeSink->write(readInto,
+                                        mBytesRead >> Format_frameBitShift(mTeeSink->format()));
+                            }
+                        }
+                    }
+                } else {
+                    // resampling
+
+                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+                    // alter output frame count as if we were expecting stereo samples
+                    if (mChannelCount == 1 && mReqChannelCount == 1) {
+                        framesOut >>= 1;
+                    }
+                    mResampler->resample(mRsmpOutBuffer, framesOut,
+                            this /* AudioBufferProvider* */);
+                    // ditherAndClamp() works as long as all buffers returned by
+                    // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
+                    if (mChannelCount == 2 && mReqChannelCount == 1) {
+                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+                        // the resampler always outputs stereo samples:
+                        // do post stereo to mono conversion
+                        downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
+                                framesOut);
+                    } else {
+                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+                    }
+
+                }
+                if (mFramestoDrop == 0) {
+                    mActiveTrack->releaseBuffer(&buffer);
+                } else {
+                    if (mFramestoDrop > 0) {
+                        mFramestoDrop -= buffer.frameCount;
+                        if (mFramestoDrop <= 0) {
+                            clearSyncStartEvent();
+                        }
+                    } else {
+                        mFramestoDrop += buffer.frameCount;
+                        if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
+                                mSyncStartEvent->isCancelled()) {
+                            ALOGW("Synced record %s, session %d, trigger session %d",
+                                  (mFramestoDrop >= 0) ? "timed out" : "cancelled",
+                                  mActiveTrack->sessionId(),
+                                  (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
+                            clearSyncStartEvent();
+                        }
+                    }
+                }
+                mActiveTrack->clearOverflow();
+            }
+            // client isn't retrieving buffers fast enough
+            else {
+                if (!mActiveTrack->setOverflow()) {
+                    nsecs_t now = systemTime();
+                    if ((now - lastWarning) > kWarningThrottleNs) {
+                        ALOGW("RecordThread: buffer overflow");
+                        lastWarning = now;
+                    }
+                }
+                // Release the processor for a while before asking for a new buffer.
+                // This will give the application more chance to read from the buffer and
+                // clear the overflow.
+                usleep(kRecordThreadSleepUs);
+            }
+        }
+        // enable changes in effect chain
+        unlockEffectChains(effectChains);
+        effectChains.clear();
+    }
+
+    standby();
+
+    {
+        Mutex::Autolock _l(mLock);
+        mActiveTrack.clear();
+        mStartStopCond.broadcast();
+    }
+
+    releaseWakeLock();
+
+    ALOGV("RecordThread %p exiting", this);
+    return false;
+}
+
+void AudioFlinger::RecordThread::standby()
+{
+    if (!mStandby) {
+        inputStandBy();
+        mStandby = true;
+    }
+}
+
+void AudioFlinger::RecordThread::inputStandBy()
+{
+    mInput->stream->common.standby(&mInput->stream->common);
+}
+
+sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
+        const sp<AudioFlinger::Client>& client,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        int sessionId,
+        IAudioFlinger::track_flags_t flags,
+        pid_t tid,
+        status_t *status)
+{
+    sp<RecordTrack> track;
+    status_t lStatus;
+
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGE("Audio driver not initialized.");
+        goto Exit;
+    }
+
+    // FIXME use flags and tid similar to createTrack_l()
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        track = new RecordTrack(this, client, sampleRate,
+                      format, channelMask, frameCount, sessionId);
+
+        if (track->getCblk() == 0) {
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
+
+        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
+        bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+                        mAudioFlinger->btNrecIsOff();
+        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
+        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
+    }
+    lStatus = NO_ERROR;
+
+Exit:
+    if (status) {
+        *status = lStatus;
+    }
+    return track;
+}
+
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
+                                           AudioSystem::sync_event_t event,
+                                           int triggerSession)
+{
+    ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
+    sp<ThreadBase> strongMe = this;
+    status_t status = NO_ERROR;
+
+    if (event == AudioSystem::SYNC_EVENT_NONE) {
+        clearSyncStartEvent();
+    } else if (event != AudioSystem::SYNC_EVENT_SAME) {
+        mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
+                                       triggerSession,
+                                       recordTrack->sessionId(),
+                                       syncStartEventCallback,
+                                       this);
+        // Sync event can be cancelled by the trigger session if the track is not in a
+        // compatible state in which case we start record immediately
+        if (mSyncStartEvent->isCancelled()) {
+            clearSyncStartEvent();
+        } else {
+            // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
+            mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
+        }
+    }
+
+    {
+        AutoMutex lock(mLock);
+        if (mActiveTrack != 0) {
+            if (recordTrack != mActiveTrack.get()) {
+                status = -EBUSY;
+            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
+                mActiveTrack->mState = TrackBase::ACTIVE;
+            }
+            return status;
+        }
+
+        recordTrack->mState = TrackBase::IDLE;
+        mActiveTrack = recordTrack;
+        mLock.unlock();
+        status_t status = AudioSystem::startInput(mId);
+        mLock.lock();
+        if (status != NO_ERROR) {
+            mActiveTrack.clear();
+            clearSyncStartEvent();
+            return status;
+        }
+        mRsmpInIndex = mFrameCount;
+        mBytesRead = 0;
+        if (mResampler != NULL) {
+            mResampler->reset();
+        }
+        mActiveTrack->mState = TrackBase::RESUMING;
+        // signal thread to start
+        ALOGV("Signal record thread");
+        mWaitWorkCV.broadcast();
+        // do not wait for mStartStopCond if exiting
+        if (exitPending()) {
+            mActiveTrack.clear();
+            status = INVALID_OPERATION;
+            goto startError;
+        }
+        mStartStopCond.wait(mLock);
+        if (mActiveTrack == 0) {
+            ALOGV("Record failed to start");
+            status = BAD_VALUE;
+            goto startError;
+        }
+        ALOGV("Record started OK");
+        return status;
+    }
+startError:
+    AudioSystem::stopInput(mId);
+    clearSyncStartEvent();
+    return status;
+}
+
+void AudioFlinger::RecordThread::clearSyncStartEvent()
+{
+    if (mSyncStartEvent != 0) {
+        mSyncStartEvent->cancel();
+    }
+    mSyncStartEvent.clear();
+    mFramestoDrop = 0;
+}
+
+void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
+{
+    sp<SyncEvent> strongEvent = event.promote();
+
+    if (strongEvent != 0) {
+        RecordThread *me = (RecordThread *)strongEvent->cookie();
+        me->handleSyncStartEvent(strongEvent);
+    }
+}
+
+void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event)
+{
+    if (event == mSyncStartEvent) {
+        // TODO: use actual buffer filling status instead of 2 buffers when info is available
+        // from audio HAL
+        mFramestoDrop = mFrameCount * 2;
+    }
+}
+
+bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) {
+    ALOGV("RecordThread::stop");
+    if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
+        return false;
+    }
+    recordTrack->mState = TrackBase::PAUSING;
+    // do not wait for mStartStopCond if exiting
+    if (exitPending()) {
+        return true;
+    }
+    mStartStopCond.wait(mLock);
+    // if we have been restarted, recordTrack == mActiveTrack.get() here
+    if (exitPending() || recordTrack != mActiveTrack.get()) {
+        ALOGV("Record stopped OK");
+        return true;
+    }
+    return false;
+}
+
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+{
+    return false;
+}
+
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+{
+#if 0   // This branch is currently dead code, but is preserved in case it will be needed in future
+    if (!isValidSyncEvent(event)) {
+        return BAD_VALUE;
+    }
+
+    int eventSession = event->triggerSession();
+    status_t ret = NAME_NOT_FOUND;
+
+    Mutex::Autolock _l(mLock);
+
+    for (size_t i = 0; i < mTracks.size(); i++) {
+        sp<RecordTrack> track = mTracks[i];
+        if (eventSession == track->sessionId()) {
+            (void) track->setSyncEvent(event);
+            ret = NO_ERROR;
+        }
+    }
+    return ret;
+#else
+    return BAD_VALUE;
+#endif
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
+{
+    track->mState = TrackBase::TERMINATED;
+    // active tracks are removed by threadLoop()
+    if (mActiveTrack != track) {
+        removeTrack_l(track);
+    }
+}
+
+void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
+{
+    mTracks.remove(track);
+    // need anything related to effects here?
+}
+
+void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    dumpTracks(fd, args);
+    dumpEffectChains(fd, args);
+}
+
+void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
+    result.append(buffer);
+
+    if (mActiveTrack != 0) {
+        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
+        result.append(buffer);
+    } else {
+        result.append("No active record client\n");
+    }
+
+    write(fd, result.string(), result.size());
+
+    dumpBase(fd, args);
+}
+
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
+    result.append(buffer);
+    RecordTrack::appendDumpHeader(result);
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<RecordTrack> track = mTracks[i];
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    if (mActiveTrack != 0) {
+        snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
+        result.append(buffer);
+        RecordTrack::appendDumpHeader(result);
+        mActiveTrack->dump(buffer, SIZE);
+        result.append(buffer);
+
+    }
+    write(fd, result.string(), result.size());
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+    size_t framesReq = buffer->frameCount;
+    size_t framesReady = mFrameCount - mRsmpInIndex;
+    int channelCount;
+
+    if (framesReady == 0) {
+        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
+        if (mBytesRead <= 0) {
+            if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
+                ALOGE("RecordThread::getNextBuffer() Error reading audio input");
+                // Force input into standby so that it tries to
+                // recover at next read attempt
+                inputStandBy();
+                usleep(kRecordThreadSleepUs);
+            }
+            buffer->raw = NULL;
+            buffer->frameCount = 0;
+            return NOT_ENOUGH_DATA;
+        }
+        mRsmpInIndex = 0;
+        framesReady = mFrameCount;
+    }
+
+    if (framesReq > framesReady) {
+        framesReq = framesReady;
+    }
+
+    if (mChannelCount == 1 && mReqChannelCount == 2) {
+        channelCount = 1;
+    } else {
+        channelCount = 2;
+    }
+    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+    buffer->frameCount = framesReq;
+    return NO_ERROR;
+}
+
+// AudioBufferProvider interface
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    mRsmpInIndex += buffer->frameCount;
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+        audio_format_t reqFormat = mFormat;
+        uint32_t reqSamplingRate = mReqSampleRate;
+        uint32_t reqChannelCount = mReqChannelCount;
+
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reqSamplingRate = value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            reqFormat = (audio_format_t) value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            reqChannelCount = popcount(value);
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be guaranteed
+            // if frame count is changed after track creation
+            if (mActiveTrack != 0) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setDevice_l(value);
+            }
+
+            // store input device and output device but do not forward output device to audio HAL.
+            // Note that status is ignored by the caller for output device
+            // (see AudioFlinger::setParameters()
+            if (audio_is_output_devices(value)) {
+                mOutDevice = value;
+                status = BAD_VALUE;
+            } else {
+                mInDevice = value;
+                // disable AEC and NS if the device is a BT SCO headset supporting those
+                // pre processings
+                if (mTracks.size() > 0) {
+                    bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+                                        mAudioFlinger->btNrecIsOff();
+                    for (size_t i = 0; i < mTracks.size(); i++) {
+                        sp<RecordTrack> track = mTracks[i];
+                        setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
+                        setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
+                    }
+                }
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
+                mAudioSource != (audio_source_t)value) {
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setAudioSource_l((audio_source_t)value);
+            }
+            mAudioSource = (audio_source_t)value;
+        }
+        if (status == NO_ERROR) {
+            status = mInput->stream->common.set_parameters(&mInput->stream->common,
+                    keyValuePair.string());
+            if (status == INVALID_OPERATION) {
+                inputStandBy();
+                status = mInput->stream->common.set_parameters(&mInput->stream->common,
+                        keyValuePair.string());
+            }
+            if (reconfig) {
+                if (status == BAD_VALUE &&
+                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
+                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common)
+                            <= (2 * reqSamplingRate)) &&
+                    popcount(mInput->stream->common.get_channels(&mInput->stream->common))
+                            <= FCC_2 &&
+                    (reqChannelCount <= FCC_2)) {
+                    status = NO_ERROR;
+                }
+                if (status == NO_ERROR) {
+                    readInputParameters();
+                    sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
+                }
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+        // already timed out waiting for the status and will never signal the condition.
+        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+    }
+    return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+    char *s;
+    String8 out_s8 = String8();
+
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return out_s8;
+    }
+
+    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
+    out_s8 = String8(s);
+    free(s);
+    return out_s8;
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = NULL;
+
+    switch (event) {
+    case AudioSystem::INPUT_OPENED:
+    case AudioSystem::INPUT_CONFIG_CHANGED:
+        desc.channels = mChannelMask;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mFrameCount;
+        desc.latency = 0;
+        param2 = &desc;
+        break;
+
+    case AudioSystem::INPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+    delete mRsmpInBuffer;
+    // mRsmpInBuffer is always assigned a new[] below
+    delete mRsmpOutBuffer;
+    mRsmpOutBuffer = NULL;
+    delete mResampler;
+    mResampler = NULL;
+
+    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
+    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
+    mChannelCount = (uint16_t)popcount(mChannelMask);
+    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
+    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
+    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
+    mFrameCount = mInputBytes / mFrameSize;
+    mNormalFrameCount = mFrameCount; // not used by record, but used by input effects
+    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
+    {
+        int channelCount;
+        // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+        // stereo to mono post process as the resampler always outputs stereo.
+        if (mChannelCount == 1 && mReqChannelCount == 2) {
+            channelCount = 1;
+        } else {
+            channelCount = 2;
+        }
+        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+        mResampler->setSampleRate(mSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+        // optmization: if mono to mono, alter input frame count as if we were inputing
+        // stereo samples
+        if (mChannelCount == 1 && mReqChannelCount == 1) {
+            mFrameCount >>= 1;
+        }
+
+    }
+    mRsmpInIndex = mFrameCount;
+}
+
+unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+{
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return 0;
+    }
+
+    return mInput->stream->get_input_frames_lost(mInput->stream);
+}
+
+uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
+{
+    Mutex::Autolock _l(mLock);
+    uint32_t result = 0;
+    if (getEffectChain_l(sessionId) != 0) {
+        result = EFFECT_SESSION;
+    }
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        if (sessionId == mTracks[i]->sessionId()) {
+            result |= TRACK_SESSION;
+            break;
+        }
+    }
+
+    return result;
+}
+
+KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
+{
+    KeyedVector<int, bool> ids;
+    Mutex::Autolock _l(mLock);
+    for (size_t j = 0; j < mTracks.size(); ++j) {
+        sp<RecordThread::RecordTrack> track = mTracks[j];
+        int sessionId = track->sessionId();
+        if (ids.indexOfKey(sessionId) < 0) {
+            ids.add(sessionId, true);
+        }
+    }
+    return ids;
+}
+
+AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
+{
+    Mutex::Autolock _l(mLock);
+    AudioStreamIn *input = mInput;
+    mInput = NULL;
+    return input;
+}
+
+// this method must always be called either with ThreadBase mLock held or inside the thread loop
+audio_stream_t* AudioFlinger::RecordThread::stream() const
+{
+    if (mInput == NULL) {
+        return NULL;
+    }
+    return &mInput->stream->common;
+}
+
+status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+    // only one chain per input thread
+    if (mEffectChains.size() != 0) {
+        return INVALID_OPERATION;
+    }
+    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
+
+    chain->setInBuffer(NULL);
+    chain->setOutBuffer(NULL);
+
+    checkSuspendOnAddEffectChain_l(chain);
+
+    mEffectChains.add(chain);
+
+    return NO_ERROR;
+}
+
+size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
+    ALOGW_IF(mEffectChains.size() != 1,
+            "removeEffectChain_l() %p invalid chain size %d on thread %p",
+            chain.get(), mEffectChains.size(), this);
+    if (mEffectChains.size() == 1) {
+        mEffectChains.removeAt(0);
+    }
+    return 0;
+}
+
+}; // namespace android