/services/audioflinger/Threads.cpp

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/*
**
** 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
#define ATRACE_TAG ATRACE_TAG_AUDIO

#include "Configuration.h"
#include <math.h>
#include <fcntl.h>
#include <sys/stat.h>
#include <cutils/properties.h>
#include <media/AudioParameter.h>
#include <media/AudioResamplerPublic.h>
#include <utils/Log.h>
#include <utils/Trace.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>
#include <audio_utils/format.h>
#include <audio_utils/minifloat.h>

// NBAIO implementations
#include <media/nbaio/AudioStreamInSource.h>
#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 "FastCapture.h"
#include "ServiceUtilities.h"
#include "SchedulingPolicyService.h"

#ifdef ADD_BATTERY_DATA
#include <media/IMediaPlayerService.h>
#include <media/IMediaDeathNotifier.h>
#endif

#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

#define max(a, b) ((a) > (b) ? (a) : (b))

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 in sendConfigEvent_l() for a status to be received
static const nsecs_t kConfigEventTimeoutNs = 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 sink buffer size, expressed in milliseconds rather than frames
static const uint32_t kMinNormalSinkBufferSizeMs = 20;
// maximum normal sink buffer size
static const uint32_t kMaxNormalSinkBufferSizeMs = 24;

// Offloaded output thread standby delay: allows track transition without going to standby
static const nsecs_t kOffloadStandbyDelayNs = seconds(1);

// 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;

// Whether to use fast capture
static const enum {
    FastCapture_Never,  // never initialize or use: for debugging only
    FastCapture_Always, // always initialize and use, even if not needed: for debugging only
    FastCapture_Static, // initialize if needed, then use all the time if initialized
} kUseFastCapture = FastCapture_Static;

// Priorities for requestPriority
static const int kPriorityAudioApp = 2;
static const int kPriorityFastMixer = 3;
static const int kPriorityFastCapture = 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 N-buffering by default, but doesn't tell us about the value of N.
// So for now we just assume that client is double-buffered for fast tracks.
// FIXME It would be better for client to tell AudioFlinger the value of N,
// so AudioFlinger could allocate the right amount of memory.
// See the client's minBufCount and mNotificationFramesAct calculations for details.

// This is the default value, if not specified by property.
static const int kFastTrackMultiplier = 2;

// The minimum and maximum allowed values
static const int kFastTrackMultiplierMin = 1;
static const int kFastTrackMultiplierMax = 2;

// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
static int sFastTrackMultiplier = kFastTrackMultiplier;

// See Thread::readOnlyHeap().
// Initially this heap is used to allocate client buffers for "fast" AudioRecord.
// Eventually it will be the single buffer that FastCapture writes into via HAL read(),
// and that all "fast" AudioRecord clients read from.  In either case, the size can be small.
static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;

// ----------------------------------------------------------------------------

static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;

static void sFastTrackMultiplierInit()
{
    char value[PROPERTY_VALUE_MAX];
    if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
        char *endptr;
        unsigned long ul = strtoul(value, &endptr, 0);
        if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
            sFastTrackMultiplier = (int) ul;
        }
    }
}

// ----------------------------------------------------------------------------

#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
#ifndef DEBUG_CPU_USAGE
                __unused
#endif
        ) {
#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, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
        // are set by PlaybackThread::readOutputParameters_l() or
        // RecordThread::readInputParameters_l()
        //FIXME: mStandby should be true here. Is this some kind of hack?
        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()
{
    // mConfigEvents should be empty, but just in case it isn't, free the memory it owns
    mConfigEvents.clear();

    // do not lock the mutex in destructor
    releaseWakeLock_l();
    if (mPowerManager != 0) {
        sp<IBinder> binder = mPowerManager->asBinder();
        binder->unlinkToDeath(mDeathRecipient);
    }
}

status_t AudioFlinger::ThreadBase::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::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);

    return sendSetParameterConfigEvent_l(keyValuePairs);
}

// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
{
    status_t status = NO_ERROR;

    mConfigEvents.add(event);
    ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType);
    mWaitWorkCV.signal();
    mLock.unlock();
    {
        Mutex::Autolock _l(event->mLock);
        while (event->mWaitStatus) {
            if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) {
                event->mStatus = TIMED_OUT;
                event->mWaitStatus = false;
            }
        }
        status = event->mStatus;
    }
    mLock.lock();
    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)
{
    sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param);
    sendConfigEvent_l(configEvent);
}

// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
{
    sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio);
    sendConfigEvent_l(configEvent);
}

// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
{
    sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair);
    return sendConfigEvent_l(configEvent);
}

status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
                                                        const struct audio_patch *patch,
                                                        audio_patch_handle_t *handle)
{
    Mutex::Autolock _l(mLock);
    sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
    status_t status = sendConfigEvent_l(configEvent);
    if (status == NO_ERROR) {
        CreateAudioPatchConfigEventData *data =
                                        (CreateAudioPatchConfigEventData *)configEvent->mData.get();
        *handle = data->mHandle;
    }
    return status;
}

status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
                                                                const audio_patch_handle_t handle)
{
    Mutex::Autolock _l(mLock);
    sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
    return sendConfigEvent_l(configEvent);
}


// post condition: mConfigEvents.isEmpty()
void AudioFlinger::ThreadBase::processConfigEvents_l()
{
    bool configChanged = false;

    while (!mConfigEvents.isEmpty()) {
        ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size());
        sp<ConfigEvent> event = mConfigEvents[0];
        mConfigEvents.removeAt(0);
        switch (event->mType) {
        case CFG_EVENT_PRIO: {
            PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
            // FIXME Need to understand why this has to be done asynchronously
            int err = requestPriority(data->mPid, data->mTid, data->mPrio,
                    true /*asynchronous*/);
            if (err != 0) {
                ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                      data->mPrio, data->mPid, data->mTid, err);
            }
        } break;
        case CFG_EVENT_IO: {
            IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
            audioConfigChanged(data->mEvent, data->mParam);
        } break;
        case CFG_EVENT_SET_PARAMETER: {
            SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
            if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
                configChanged = true;
            }
        } break;
        case CFG_EVENT_CREATE_AUDIO_PATCH: {
            CreateAudioPatchConfigEventData *data =
                                            (CreateAudioPatchConfigEventData *)event->mData.get();
            event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
        } break;
        case CFG_EVENT_RELEASE_AUDIO_PATCH: {
            ReleaseAudioPatchConfigEventData *data =
                                            (ReleaseAudioPatchConfigEventData *)event->mData.get();
            event->mStatus = releaseAudioPatch_l(data->mHandle);
        } break;
        default:
            ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
            break;
        }
        {
            Mutex::Autolock _l(event->mLock);
            if (event->mWaitStatus) {
                event->mWaitStatus = false;
                event->mCond.signal();
            }
        }
        ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
    }

    if (configChanged) {
        cacheParameters_l();
    }
}

String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
    String8 s;
    if (output) {
        if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
        if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
        if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
        if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
        if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
        if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
        if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
        if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
        if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
        if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
        if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
        if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
        if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown,  ");
    } else {
        if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
        if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
        if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
        if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
        if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
        if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
        if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
        if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
        if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
        if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
        if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
        if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
        if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
        if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
        if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown,  ");
    }
    int len = s.length();
    if (s.length() > 2) {
        char *str = s.lockBuffer(len);
        s.unlockBuffer(len - 2);
    }
    return s;
}

void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
{
    const size_t SIZE = 256;
    char buffer[SIZE];
    String8 result;

    bool locked = AudioFlinger::dumpTryLock(mLock);
    if (!locked) {
        dprintf(fd, "thread %p maybe dead locked\n", this);
    }

    dprintf(fd, "  I/O handle: %d\n", mId);
    dprintf(fd, "  TID: %d\n", getTid());
    dprintf(fd, "  Standby: %s\n", mStandby ? "yes" : "no");
    dprintf(fd, "  Sample rate: %u\n", mSampleRate);
    dprintf(fd, "  HAL frame count: %zu\n", mFrameCount);
    dprintf(fd, "  HAL buffer size: %u bytes\n", mBufferSize);
    dprintf(fd, "  Channel Count: %u\n", mChannelCount);
    dprintf(fd, "  Channel Mask: 0x%08x (%s)\n", mChannelMask,
            channelMaskToString(mChannelMask, mType != RECORD).string());
    dprintf(fd, "  Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
    dprintf(fd, "  Frame size: %zu\n", mFrameSize);
    dprintf(fd, "  Pending config events:");
    size_t numConfig = mConfigEvents.size();
    if (numConfig) {
        for (size_t i = 0; i < numConfig; i++) {
            mConfigEvents[i]->dump(buffer, SIZE);
            dprintf(fd, "\n    %s", buffer);
        }
        dprintf(fd, "\n");
    } else {
        dprintf(fd, " none\n");
    }

    if (locked) {
        mLock.unlock();
    }
}

void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
{
    const size_t SIZE = 256;
    char buffer[SIZE];
    String8 result;

    size_t numEffectChains = mEffectChains.size();
    snprintf(buffer, SIZE, "  %zu Effect Chains\n", numEffectChains);
    write(fd, buffer, strlen(buffer));

    for (size_t i = 0; i < numEffectChains; ++i) {
        sp<EffectChain> chain = mEffectChains[i];
        if (chain != 0) {
            chain->dump(fd, args);
        }
    }
}

void AudioFlinger::ThreadBase::acquireWakeLock(int uid)
{
    Mutex::Autolock _l(mLock);
    acquireWakeLock_l(uid);
}

String16 AudioFlinger::ThreadBase::getWakeLockTag()
{
    switch (mType) {
        case MIXER:
            return String16("AudioMix");
        case DIRECT:
            return String16("AudioDirectOut");
        case DUPLICATING:
            return String16("AudioDup");
        case RECORD:
            return String16("AudioIn");
        case OFFLOAD:
            return String16("AudioOffload");
        default:
            ALOG_ASSERT(false);
            return String16("AudioUnknown");
    }
}

void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
{
    getPowerManager_l();
    if (mPowerManager != 0) {
        sp<IBinder> binder = new BBinder();
        status_t status;
        if (uid >= 0) {
            status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
                    binder,
                    getWakeLockTag(),
                    String16("media"),
                    uid,
                    true /* FIXME force oneway contrary to .aidl */);
        } else {
            status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
                    binder,
                    getWakeLockTag(),
                    String16("media"),
                    true /* FIXME force oneway contrary to .aidl */);
        }
        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,
                    true /* FIXME force oneway contrary to .aidl */);
        }
        mWakeLockToken.clear();
    }
}

void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) {
    Mutex::Autolock _l(mLock);
    updateWakeLockUids_l(uids);
}

void AudioFlinger::ThreadBase::getPowerManager_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);
        }
    }
}

void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {

    getPowerManager_l();
    if (mWakeLockToken == NULL) {
        ALOGE("no wake lock to update!");
        return;
    }
    if (mPowerManager != 0) {
        sp<IBinder> binder = new BBinder();
        status_t status;
        status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
                    true /* FIXME force oneway contrary to .aidl */);
        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
    }
}

void AudioFlinger::ThreadBase::clearPowerManager()
{
    Mutex::Autolock _l(mLock);
    releaseWakeLock_l();
    mPowerManager.clear();
}

void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
{
    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;
    }

    // Reject any effect on Direct output threads for now, since the format of
    // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
    if (mType == DIRECT) {
        ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s",
                desc->name, mName);
        lStatus = BAD_VALUE;
        goto Exit;
    }

    // Reject any effect on mixer or duplicating multichannel sinks.
    // TODO: fix both format and multichannel issues with effects.
    if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) {
        ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads",
                desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING");
        lStatus = BAD_VALUE;
        goto Exit;
    }

    // Allow global effects only on offloaded and mixer threads
    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
        switch (mType) {
        case MIXER:
        case OFFLOAD:
            break;
        case DIRECT:
        case DUPLICATING:
        case RECORD:
        default:
            ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName);
            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;
            }
            effect->setOffloaded(mType == OFFLOAD, mId);

            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 = handle->initCheck();
        if (lStatus == OK) {
            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();
    }

    *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;

    ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(),
             "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x",
                    this, effect->desc().name, effect->desc().flags);

    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;
    }

    effect->setOffloaded(mType == OFFLOAD, mId);

    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::getAudioPortConfig(struct audio_port_config *config)
{
    config->type = AUDIO_PORT_TYPE_MIX;
    config->ext.mix.handle = mId;
    config->sample_rate = mSampleRate;
    config->format = mFormat;
    config->channel_mask = mChannelMask;
    config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
                            AUDIO_PORT_CONFIG_FORMAT;
}


// ----------------------------------------------------------------------------
//      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),
        mNormalFrameCount(0), mSinkBuffer(NULL),
        mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
        mMixerBuffer(NULL),
        mMixerBufferSize(0),
        mMixerBufferFormat(AUDIO_FORMAT_INVALID),
        mMixerBufferValid(false),
        mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
        mEffectBuffer(NULL),
        mEffectBufferSize(0),
        mEffectBufferFormat(AUDIO_FORMAT_INVALID),
        mEffectBufferValid(false),
        mSuspended(0), mBytesWritten(0),
        mActiveTracksGeneration(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),
        mBytesRemaining(0),
        mCurrentWriteLength(0),
        mUseAsyncWrite(false),
        mWriteAckSequence(0),
        mDrainSequence(0),
        mSignalPending(false),
        mScreenState(AudioFlinger::mScreenState),
        // index 0 is reserved for normal mixer's submix
        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1),
        mHwSupportsPause(false), mHwPaused(false), mFlushPending(false),
        // mLatchD, mLatchQ,
        mLatchDValid(false), mLatchQValid(false)
{
    snprintf(mName, kNameLength, "AudioOut_%X", id);
    mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);

    // 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_l();

    // ++ operator does not compile
    for (audio_stream_type_t stream = AUDIO_STREAM_MIN; 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);
    }
}

AudioFlinger::PlaybackThread::~PlaybackThread()
{
    mAudioFlinger->unregisterWriter(mNBLogWriter);
    free(mSinkBuffer);
    free(mMixerBuffer);
    free(mEffectBuffer);
}

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 __unused)
{
    const size_t SIZE = 256;
    char buffer[SIZE];
    String8 result;

    result.appendFormat("  Stream volumes in dB: ");
    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();

    // These values are "raw"; they will wrap around.  See prepareTracks_l() for a better way.
    FastTrackUnderruns underruns = getFastTrackUnderruns(0);
    dprintf(fd, "  Normal mixer raw underrun counters: partial=%u empty=%u\n",
            underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);

    size_t numtracks = mTracks.size();
    size_t numactive = mActiveTracks.size();
    dprintf(fd, "  %d Tracks", numtracks);
    size_t numactiveseen = 0;
    if (numtracks) {
        dprintf(fd, " of which %d are active\n", numactive);
        Track::appendDumpHeader(result);
        for (size_t i = 0; i < numtracks; ++i) {
            sp<Track> track = mTracks[i];
            if (track != 0) {
                bool active = mActiveTracks.indexOf(track) >= 0;
                if (active) {
                    numactiveseen++;
                }
                track->dump(buffer, SIZE, active);
                result.append(buffer);
            }
        }
    } else {
        result.append("\n");
    }
    if (numactiveseen != numactive) {
        // some tracks in the active list were not in the tracks list
        snprintf(buffer, SIZE, "  The following tracks are in the active list but"
                " not in the track list\n");
        result.append(buffer);
        Track::appendDumpHeader(result);
        for (size_t i = 0; i < numactive; ++i) {
            sp<Track> track = mActiveTracks[i].promote();
            if (track != 0 && mTracks.indexOf(track) < 0) {
                track->dump(buffer, SIZE, true);
                result.append(buffer);
            }
        }
    }

    write(fd, result.string(), result.size());
}

void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
    dprintf(fd, "\nOutput thread %p:\n", this);
    dprintf(fd, "  Normal frame count: %zu\n", mNormalFrameCount);
    dprintf(fd, "  Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
    dprintf(fd, "  Total writes: %d\n", mNumWrites);
    dprintf(fd, "  Delayed writes: %d\n", mNumDelayedWrites);
    dprintf(fd, "  Blocked in write: %s\n", mInWrite ? "yes" : "no");
    dprintf(fd, "  Suspend count: %d\n", mSuspended);
    dprintf(fd, "  Sink buffer : %p\n", mSinkBuffer);
    dprintf(fd, "  Mixer buffer: %p\n", mMixerBuffer);
    dprintf(fd, "  Effect buffer: %p\n", mEffectBuffer);
    dprintf(fd, "  Fast track availMask=%#x\n", mFastTrackAvailMask);

    dumpBase(fd, args);
}

// Thread virtuals

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 *pFrameCount,
        const sp<IMemory>& sharedBuffer,
        int sessionId,
        IAudioFlinger::track_flags_t *flags,
        pid_t tid,
        int uid,
        status_t *status)
{
    size_t frameCount = *pFrameCount;
    sp<Track> track;
    status_t lStatus;

    bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != …