xref: /frameworks/av/services/audioflinger/Threads.cpp

/*
**
** 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 <linux/futex.h>
#include <sys/stat.h>
#include <sys/syscall.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/conversion.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 <mediautils/BatteryNotifier.h>

#include <powermanager/PowerManager.h>

#include "AudioFlinger.h"
#include "AudioMixer.h"
#include "BufferProviders.h"
#include "FastMixer.h"
#include "FastCapture.h"
#include "ServiceUtilities.h"
#include "mediautils/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

#include "AutoPark.h"

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

// 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

// TODO: Move these macro/inlines to a header file.
#define max(a, b) ((a) > (b) ? (a) : (b))
template <typename T>
static inline T min(const T& a, const T& b)
{
    return a < b ? a : b;
}

#ifndef ARRAY_SIZE
#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[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 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
// FIXME This should be based on experimentally observed scheduling jitter
static const uint32_t kMinNormalSinkBufferSizeMs = 20;
// maximum normal sink buffer size
static const uint32_t kMaxNormalSinkBufferSizeMs = 24;

// minimum capture buffer size in milliseconds to _not_ need a fast capture thread
// FIXME This should be based on experimentally observed scheduling jitter
static const uint32_t kMinNormalCaptureBufferSizeMs = 12;

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

// Direct output thread minimum sleep time in idle or active(underrun) state
static const nsecs_t kDirectMinSleepTimeUs = 10000;


// 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() has an in/out parameter 'pFrameCount' for the total size of the
// track buffer in shared memory.  Zero on input means to use a default value.  For fast tracks,
// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'.

// 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

// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset
struct {
    // call when you acquire a partial wakelock
    void acquire(const sp<IBinder> &wakeLockToken) {
        pthread_mutex_lock(&mLock);
        if (wakeLockToken.get() == nullptr) {
            adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
        } else {
            if (mCount == 0) {
                adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
            }
            ++mCount;
        }
        pthread_mutex_unlock(&mLock);
    }

    // call when you release a partial wakelock.
    void release(const sp<IBinder> &wakeLockToken) {
        if (wakeLockToken.get() == nullptr) {
            return;
        }
        pthread_mutex_lock(&mLock);
        if (--mCount < 0) {
            ALOGE("negative wakelock count");
            mCount = 0;
        }
        pthread_mutex_unlock(&mLock);
    }

    // retrieves the boottime timebase offset from monotonic.
    int64_t getBoottimeOffset() {
        pthread_mutex_lock(&mLock);
        int64_t boottimeOffset = mBoottimeOffset;
        pthread_mutex_unlock(&mLock);
        return boottimeOffset;
    }

    // Adjusts the timebase offset between TIMEBASE_MONOTONIC
    // and the selected timebase.
    // Currently only TIMEBASE_BOOTTIME is allowed.
    //
    // This only needs to be called upon acquiring the first partial wakelock
    // after all other partial wakelocks are released.
    //
    // We do an empirical measurement of the offset rather than parsing
    // /proc/timer_list since the latter is not a formal kernel ABI.
    static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) {
        int clockbase;
        switch (timebase) {
        case ExtendedTimestamp::TIMEBASE_BOOTTIME:
            clockbase = SYSTEM_TIME_BOOTTIME;
            break;
        default:
            LOG_ALWAYS_FATAL("invalid timebase %d", timebase);
            break;
        }
        // try three times to get the clock offset, choose the one
        // with the minimum gap in measurements.
        const int tries = 3;
        nsecs_t bestGap, measured;
        for (int i = 0; i < tries; ++i) {
            const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC);
            const nsecs_t tbase = systemTime(clockbase);
            const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC);
            const nsecs_t gap = tmono2 - tmono;
            if (i == 0 || gap < bestGap) {
                bestGap = gap;
                measured = tbase - ((tmono + tmono2) >> 1);
            }
        }

        // to avoid micro-adjusting, we don't change the timebase
        // unless it is significantly different.
        //
        // Assumption: It probably takes more than toleranceNs to
        // suspend and resume the device.
        static int64_t toleranceNs = 10000; // 10 us
        if (llabs(*offset - measured) > toleranceNs) {
            ALOGV("Adjusting timebase offset old: %lld  new: %lld",
                    (long long)*offset, (long long)measured);
            *offset = measured;
        }
    }

    pthread_mutex_t mLock;
    int32_t mCount;
    int64_t mBoottimeOffset;
} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization

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

// static
const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
{
    switch (type) {
    case MIXER:
        return "MIXER";
    case DIRECT:
        return "DIRECT";
    case DUPLICATING:
        return "DUPLICATING";
    case RECORD:
        return "RECORD";
    case OFFLOAD:
        return "OFFLOAD";
    default:
        return "unknown";
    }
}

String8 devicesToString(audio_devices_t devices)
{
    static const struct mapping {
        audio_devices_t mDevices;
        const char *    mString;
    } mappingsOut[] = {
        {AUDIO_DEVICE_OUT_EARPIECE,         "EARPIECE"},
        {AUDIO_DEVICE_OUT_SPEAKER,          "SPEAKER"},
        {AUDIO_DEVICE_OUT_WIRED_HEADSET,    "WIRED_HEADSET"},
        {AUDIO_DEVICE_OUT_WIRED_HEADPHONE,  "WIRED_HEADPHONE"},
        {AUDIO_DEVICE_OUT_BLUETOOTH_SCO,    "BLUETOOTH_SCO"},
        {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET,    "BLUETOOTH_SCO_HEADSET"},
        {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT,     "BLUETOOTH_SCO_CARKIT"},
        {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,           "BLUETOOTH_A2DP"},
        {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"},
        {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER,   "BLUETOOTH_A2DP_SPEAKER"},
        {AUDIO_DEVICE_OUT_AUX_DIGITAL,      "AUX_DIGITAL"},
        {AUDIO_DEVICE_OUT_HDMI,             "HDMI"},
        {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"},
        {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"},
        {AUDIO_DEVICE_OUT_USB_ACCESSORY,    "USB_ACCESSORY"},
        {AUDIO_DEVICE_OUT_USB_DEVICE,       "USB_DEVICE"},
        {AUDIO_DEVICE_OUT_TELEPHONY_TX,     "TELEPHONY_TX"},
        {AUDIO_DEVICE_OUT_LINE,             "LINE"},
        {AUDIO_DEVICE_OUT_HDMI_ARC,         "HDMI_ARC"},
        {AUDIO_DEVICE_OUT_SPDIF,            "SPDIF"},
        {AUDIO_DEVICE_OUT_FM,               "FM"},
        {AUDIO_DEVICE_OUT_AUX_LINE,         "AUX_LINE"},
        {AUDIO_DEVICE_OUT_SPEAKER_SAFE,     "SPEAKER_SAFE"},
        {AUDIO_DEVICE_OUT_IP,               "IP"},
        {AUDIO_DEVICE_OUT_BUS,              "BUS"},
        {AUDIO_DEVICE_NONE,                 "NONE"},       // must be last
    }, mappingsIn[] = {
        {AUDIO_DEVICE_IN_COMMUNICATION,     "COMMUNICATION"},
        {AUDIO_DEVICE_IN_AMBIENT,           "AMBIENT"},
        {AUDIO_DEVICE_IN_BUILTIN_MIC,       "BUILTIN_MIC"},
        {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"},
        {AUDIO_DEVICE_IN_WIRED_HEADSET,     "WIRED_HEADSET"},
        {AUDIO_DEVICE_IN_AUX_DIGITAL,       "AUX_DIGITAL"},
        {AUDIO_DEVICE_IN_VOICE_CALL,        "VOICE_CALL"},
        {AUDIO_DEVICE_IN_TELEPHONY_RX,      "TELEPHONY_RX"},
        {AUDIO_DEVICE_IN_BACK_MIC,          "BACK_MIC"},
        {AUDIO_DEVICE_IN_REMOTE_SUBMIX,     "REMOTE_SUBMIX"},
        {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"},
        {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"},
        {AUDIO_DEVICE_IN_USB_ACCESSORY,     "USB_ACCESSORY"},
        {AUDIO_DEVICE_IN_USB_DEVICE,        "USB_DEVICE"},
        {AUDIO_DEVICE_IN_FM_TUNER,          "FM_TUNER"},
        {AUDIO_DEVICE_IN_TV_TUNER,          "TV_TUNER"},
        {AUDIO_DEVICE_IN_LINE,              "LINE"},
        {AUDIO_DEVICE_IN_SPDIF,             "SPDIF"},
        {AUDIO_DEVICE_IN_BLUETOOTH_A2DP,    "BLUETOOTH_A2DP"},
        {AUDIO_DEVICE_IN_LOOPBACK,          "LOOPBACK"},
        {AUDIO_DEVICE_IN_IP,                "IP"},
        {AUDIO_DEVICE_IN_BUS,               "BUS"},
        {AUDIO_DEVICE_NONE,                 "NONE"},        // must be last
    };
    String8 result;
    audio_devices_t allDevices = AUDIO_DEVICE_NONE;
    const mapping *entry;
    if (devices & AUDIO_DEVICE_BIT_IN) {
        devices &= ~AUDIO_DEVICE_BIT_IN;
        entry = mappingsIn;
    } else {
        entry = mappingsOut;
    }
    for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) {
        allDevices = (audio_devices_t) (allDevices | entry->mDevices);
        if (devices & entry->mDevices) {
            if (!result.isEmpty()) {
                result.append("|");
            }
            result.append(entry->mString);
        }
    }
    if (devices & ~allDevices) {
        if (!result.isEmpty()) {
            result.append("|");
        }
        result.appendFormat("0x%X", devices & ~allDevices);
    }
    if (result.isEmpty()) {
        result.append(entry->mString);
    }
    return result;
}

String8 inputFlagsToString(audio_input_flags_t flags)
{
    static const struct mapping {
        audio_input_flags_t     mFlag;
        const char *            mString;
    } mappings[] = {
        {AUDIO_INPUT_FLAG_FAST,             "FAST"},
        {AUDIO_INPUT_FLAG_HW_HOTWORD,       "HW_HOTWORD"},
        {AUDIO_INPUT_FLAG_RAW,              "RAW"},
        {AUDIO_INPUT_FLAG_SYNC,             "SYNC"},
        {AUDIO_INPUT_FLAG_NONE,             "NONE"},        // must be last
    };
    String8 result;
    audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE;
    const mapping *entry;
    for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) {
        allFlags = (audio_input_flags_t) (allFlags | entry->mFlag);
        if (flags & entry->mFlag) {
            if (!result.isEmpty()) {
                result.append("|");
            }
            result.append(entry->mString);
        }
    }
    if (flags & ~allFlags) {
        if (!result.isEmpty()) {
            result.append("|");
        }
        result.appendFormat("0x%X", flags & ~allFlags);
    }
    if (result.isEmpty()) {
        result.append(entry->mString);
    }
    return result;
}

String8 outputFlagsToString(audio_output_flags_t flags)
{
    static const struct mapping {
        audio_output_flags_t    mFlag;
        const char *            mString;
    } mappings[] = {
        {AUDIO_OUTPUT_FLAG_DIRECT,          "DIRECT"},
        {AUDIO_OUTPUT_FLAG_PRIMARY,         "PRIMARY"},
        {AUDIO_OUTPUT_FLAG_FAST,            "FAST"},
        {AUDIO_OUTPUT_FLAG_DEEP_BUFFER,     "DEEP_BUFFER"},
        {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"},
        {AUDIO_OUTPUT_FLAG_NON_BLOCKING,    "NON_BLOCKING"},
        {AUDIO_OUTPUT_FLAG_HW_AV_SYNC,      "HW_AV_SYNC"},
        {AUDIO_OUTPUT_FLAG_RAW,             "RAW"},
        {AUDIO_OUTPUT_FLAG_SYNC,            "SYNC"},
        {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"},
        {AUDIO_OUTPUT_FLAG_NONE,            "NONE"},        // must be last
    };
    String8 result;
    audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
    const mapping *entry;
    for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
        allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
        if (flags & entry->mFlag) {
            if (!result.isEmpty()) {
                result.append("|");
            }
            result.append(entry->mString);
        }
    }
    if (flags & ~allFlags) {
        if (!result.isEmpty()) {
            result.append("|");
        }
        result.appendFormat("0x%X", flags & ~allFlags);
    }
    if (result.isEmpty()) {
        result.append(entry->mString);
    }
    return result;
}

const char *sourceToString(audio_source_t source)
{
    switch (source) {
    case AUDIO_SOURCE_DEFAULT:              return "default";
    case AUDIO_SOURCE_MIC:                  return "mic";
    case AUDIO_SOURCE_VOICE_UPLINK:         return "voice uplink";
    case AUDIO_SOURCE_VOICE_DOWNLINK:       return "voice downlink";
    case AUDIO_SOURCE_VOICE_CALL:           return "voice call";
    case AUDIO_SOURCE_CAMCORDER:            return "camcorder";
    case AUDIO_SOURCE_VOICE_RECOGNITION:    return "voice recognition";
    case AUDIO_SOURCE_VOICE_COMMUNICATION:  return "voice communication";
    case AUDIO_SOURCE_REMOTE_SUBMIX:        return "remote submix";
    case AUDIO_SOURCE_UNPROCESSED:          return "unprocessed";
    case AUDIO_SOURCE_FM_TUNER:             return "FM tuner";
    case AUDIO_SOURCE_HOTWORD:              return "hotword";
    default:                                return "unknown";
    }
}

AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
        audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady)
    :   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),
        mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE),
        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
        // mName will be set by concrete (non-virtual) subclass
        mDeathRecipient(new PMDeathRecipient(this)),
        mSystemReady(systemReady),
        mNotifiedBatteryStart(false)
{
    memset(&mPatch, 0, sizeof(struct audio_patch));
}

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 = IInterface::asBinder(mPowerManager);
        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)
{
    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;

    if (event->mRequiresSystemReady && !mSystemReady) {
        event->mWaitStatus = false;
        mPendingConfigEvents.add(event);
        return status;
    }
    mConfigEvents.add(event);
    ALOGV("sendConfigEvent_l() num events %zu 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(audio_io_config_event event, pid_t pid)
{
    Mutex::Autolock _l(mLock);
    sendIoConfigEvent_l(event, pid);
}

// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid)
{
    sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid);
    sendConfigEvent_l(configEvent);
}

void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio)
{
    Mutex::Autolock _l(mLock);
    sendPrioConfigEvent_l(pid, tid, prio);
}

// 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;
    AudioParameter param(keyValuePair);
    int value;
    if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) {
        setMasterMono_l(value != 0);
        if (param.size() == 1) {
            return NO_ERROR; // should be a solo parameter - we don't pass down
        }
        param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT));
        configEvent = new SetParameterConfigEvent(param.toString());
    } else {
        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 %zu", 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();
            ioConfigChanged(data->mEvent, data->mPid);
        } 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;
    const audio_channel_representation_t representation =
            audio_channel_mask_get_representation(mask);

    switch (representation) {
    case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
        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,  ");
        }
        const int len = s.length();
        if (len > 2) {
            (void) s.lockBuffer(len);      // needed?
            s.unlockBuffer(len - 2);       // remove trailing ", "
        }
        return s;
    }
    case AUDIO_CHANNEL_REPRESENTATION_INDEX:
        s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask));
        return s;
    default:
        s.appendFormat("unknown mask, representation:%d  bits:%#x",
                representation, audio_channel_mask_get_bits(mask));
        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 may be deadlocked\n", this);
    }

    dprintf(fd, "  Thread name: %s\n", mThreadName);
    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 Hz\n", mSampleRate);
    dprintf(fd, "  HAL frame count: %zu\n", mFrameCount);
    dprintf(fd, "  HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
    dprintf(fd, "  HAL buffer size: %zu 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, "  Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
    dprintf(fd, "  Processing frame size: %zu bytes\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");
    }
    dprintf(fd, "  Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string());
    dprintf(fd, "  Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string());
    dprintf(fd, "  Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));

    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("audioserver"),
                    uid,
                    true /* FIXME force oneway contrary to .aidl */);
        } else {
            status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
                    binder,
                    getWakeLockTag(),
                    String16("audioserver"),
                    true /* FIXME force oneway contrary to .aidl */);
        }
        if (status == NO_ERROR) {
            mWakeLockToken = binder;
        }
        ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
    }

    if (!mNotifiedBatteryStart) {
        BatteryNotifier::getInstance().noteStartAudio();
        mNotifiedBatteryStart = true;
    }
    gBoottime.acquire(mWakeLockToken);
    mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] =
            gBoottime.getBoottimeOffset();
}

void AudioFlinger::ThreadBase::releaseWakeLock()
{
    Mutex::Autolock _l(mLock);
    releaseWakeLock_l();
}

void AudioFlinger::ThreadBase::releaseWakeLock_l()
{
    gBoottime.release(mWakeLockToken);
    if (mWakeLockToken != 0) {
        ALOGV("releaseWakeLock_l() %s", mThreadName);
        if (mPowerManager != 0) {
            mPowerManager->releaseWakeLock(mWakeLockToken, 0,
                    true /* FIXME force oneway contrary to .aidl */);
        }
        mWakeLockToken.clear();
    }

    if (mNotifiedBatteryStart) {
        BatteryNotifier::getInstance().noteStopAudio();
        mNotifiedBatteryStart = false;
    }
}

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

void AudioFlinger::ThreadBase::getPowerManager_l() {
    if (mSystemReady && 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", mThreadName);
        } else {
            mPowerManager = interface_cast<IPowerManager>(binder);
            binder->linkToDeath(mDeathRecipient);
        }
    }
}

void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
    getPowerManager_l();
    if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called.
        if (mSystemReady) {
            ALOGE("no wake lock to update, but system ready!");
        } else {
            ALOGW("no wake lock to update, system not ready yet");
        }
        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("updateWakeLockUids_l() %s status %d", mThreadName, 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, audio_session_t sessionId)
{
    Mutex::Autolock _l(mLock);
    setEffectSuspended_l(type, suspend, sessionId);
}

void AudioFlinger::ThreadBase::setEffectSuspended_l(
        const effect_uuid_t *type, bool suspend, audio_session_t 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,
                                                         audio_session_t 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,
                                                            audio_session_t sessionId)
{
    Mutex::Autolock _l(mLock);
    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
}

void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
                                                            bool enabled,
                                                            audio_session_t 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);
    }
}

// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::RecordThread::checkEffectCompatibility_l(
        const effect_descriptor_t *desc, audio_session_t sessionId)
{
    // No global effect sessions on record threads
    if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
        ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s",
                desc->name, mThreadName);
        return BAD_VALUE;
    }
    // only pre processing effects on record thread
    if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) {
        ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s",
                desc->name, mThreadName);
        return BAD_VALUE;
    }
    audio_input_flags_t flags = mInput->flags;
    if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) {
        if (flags & AUDIO_INPUT_FLAG_RAW) {
            ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode",
                  desc->name, mThreadName);
            return BAD_VALUE;
        }
        if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
            ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode",
                  desc->name, mThreadName);
            return BAD_VALUE;
        }
    }
    return NO_ERROR;
}

// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l(
        const effect_descriptor_t *desc, audio_session_t sessionId)
{
    // no preprocessing on playback threads
    if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) {
        ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback"
                " thread %s", desc->name, mThreadName);
        return BAD_VALUE;
    }

    switch (mType) {
    case MIXER: {
        // Reject any effect on mixer multichannel sinks.
        // TODO: fix both format and multichannel issues with effects.
        if (mChannelCount != FCC_2) {
            ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER"
                    " thread %s", desc->name, mChannelCount, mThreadName);
            return BAD_VALUE;
        }
        audio_output_flags_t flags = mOutput->flags;
        if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) {
            if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
                // global effects are applied only to non fast tracks if they are SW
                if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
                    break;
                }
            } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
                // only post processing on output stage session
                if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
                    ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed"
                            " on output stage session", desc->name);
                    return BAD_VALUE;
                }
            } else {
                // no restriction on effects applied on non fast tracks
                if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) {
                    break;
                }
            }
            if (flags & AUDIO_OUTPUT_FLAG_RAW) {
                ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode",
                      desc->name);
                return BAD_VALUE;
            }
            if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
                ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread"
                        " in fast mode", desc->name);
                return BAD_VALUE;
            }
        }
    } break;
    case OFFLOAD:
        // nothing actionable on offload threads, if the effect:
        //   - is offloadable: the effect can be created
        //   - is NOT offloadable: the effect should still be created, but EffectHandle::enable()
        //     will take care of invalidating the tracks of the thread
        break;
    case DIRECT:
        // 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).
        ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s",
                desc->name, mThreadName);
        return BAD_VALUE;
    case DUPLICATING:
        // Reject any effect on mixer multichannel sinks.
        // TODO: fix both format and multichannel issues with effects.
        if (mChannelCount != FCC_2) {
            ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)"
                    " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName);
            return BAD_VALUE;
        }
        if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) {
            ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING"
                    " thread %s", desc->name, mThreadName);
            return BAD_VALUE;
        }
        if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
            ALOGW("checkEffectCompatibility_l(): post processing effect %s on"
                    " DUPLICATING thread %s", desc->name, mThreadName);
            return BAD_VALUE;
        }
        if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) {
            ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on"
                    " DUPLICATING thread %s", desc->name, mThreadName);
            return BAD_VALUE;
        }
        break;
    default:
        LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
    }

    return NO_ERROR;
}

// 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,
        audio_session_t 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;
    }

    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);

    { // scope for mLock
        Mutex::Autolock _l(mLock);

        lStatus = checkEffectCompatibility_l(desc, sessionId);
        if (lStatus != NO_ERROR) {
            goto Exit;
        }

        // 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) {
            audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
            // 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(audio_session_t sessionId,
        int effectId)
{
    Mutex::Autolock _l(mLock);
    return getEffect_l(sessionId, effectId);
}

sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t 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
    audio_session_t 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(audio_session_t sessionId)
{
    Mutex::Autolock _l(mLock);
    return getEffectChain_l(sessionId);
}

sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t 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;
}

void AudioFlinger::ThreadBase::systemReady()
{
    Mutex::Autolock _l(mLock);
    if (mSystemReady) {
        return;
    }
    mSystemReady = true;

    for (size_t i = 0; i < mPendingConfigEvents.size(); i++) {
        sendConfigEvent_l(mPendingConfigEvents.editItemAt(i));
    }
    mPendingConfigEvents.clear();
}


// ----------------------------------------------------------------------------
//      Playback
// ----------------------------------------------------------------------------

AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
                                             AudioStreamOut* output,
                                             audio_io_handle_t id,
                                             audio_devices_t device,
                                             type_t type,
                                             bool systemReady)
    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady),
        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),
        mFramesWritten(0),
        mSuspendedFrames(0),
        mActiveTracksGeneration(0),
        // mStreamTypes[] initialized in constructor body
        mOutput(output),
        mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
        mMixerStatus(MIXER_IDLE),
        mMixerStatusIgnoringFastTracks(MIXER_IDLE),
        mStandbyDelayNs(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::sMaxFastTracks) - 1) & ~1),
        mHwSupportsPause(false), mHwPaused(false), mFlushPending(false)
{
    snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
    mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);

    // 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, "  %zu Tracks", numtracks);
    size_t numactiveseen = 0;
    if (numtracks) {
        dprintf(fd, " of which %zu 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 type %d (%s):\n", this, type(), threadTypeToString(type()));

    dumpBase(fd, args);

    dprintf(fd, "  Normal frame count: %zu\n", mNormalFrameCount);
    dprintf(fd, "  Last write occurred (msecs): %llu\n",
            (unsigned long long) 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);
    dprintf(fd, "  Standby delay ns=%lld\n", (long long)mStandbyDelayNs);
    AudioStreamOut *output = mOutput;
    audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
    String8 flagsAsString = outputFlagsToString(flags);
    dprintf(fd, "  AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
}

// Thread virtuals

void AudioFlinger::PlaybackThread::onFirstRef()
{
    run(mThreadName, 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,
        audio_session_t sessionId,
        audio_output_flags_t *flags,
        pid_t tid,
        int uid,
        status_t *status)
{
    size_t frameCount = *pFrameCount;
    sp<Track> track;
    status_t lStatus;
    audio_output_flags_t outputFlags = mOutput->flags;

    // special case for FAST flag considered OK if fast mixer is present
    if (hasFastMixer()) {
        outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST);
    }

    // Check if requested flags are compatible with output stream flags
    if ((*flags & outputFlags) != *flags) {
        ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)",
              *flags, outputFlags);
        *flags = (audio_output_flags_t)(*flags & outputFlags);
    }

    // client expresses a preference for FAST, but we get the final say
    if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
      if (
            // PCM data
            audio_is_linear_pcm(format) &&
            // TODO: extract as a data library function that checks that a computationally
            // expensive downmixer is not required: isFastOutputChannelConversion()
            (channelMask == mChannelMask ||
                    mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
                    (channelMask == AUDIO_CHANNEL_OUT_MONO
                            /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
            // hardware sample rate
            (sampleRate == mSampleRate) &&
            // 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?
        ) {
        // static tracks can have any nonzero framecount, streaming tracks check against minimum.
        if (sharedBuffer == 0) {
            // read the fast track multiplier property the first time it is needed
            int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
            if (ok != 0) {
                ALOGE("%s pthread_once failed: %d", __func__, ok);
            }
            frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
        }

        // check compatibility with audio effects.
        { // scope for mLock
            Mutex::Autolock _l(mLock);
            // do not accept RAW flag if post processing are present. Note that post processing on
            // a fast mixer are necessarily hardware
            sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE);
            if (chain != 0) {
                ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0,
                        "AUDIO_OUTPUT_FLAG_RAW denied: post processing effect present");
                *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW);
            }
            // Do not accept FAST flag if software global effects are present
            chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
            if (chain != 0) {
                ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0,
                        "AUDIO_OUTPUT_FLAG_RAW denied: global effect present");
                *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW);
                if (chain->hasSoftwareEffect()) {
                    ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software global effect present");
                    *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
                }
            }
            // Do not accept FAST flag if the session has software effects
            chain = getEffectChain_l(sessionId);
            if (chain != 0) {
                ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0,
                        "AUDIO_OUTPUT_FLAG_RAW denied: effect present on session");
                *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW);
                if (chain->hasSoftwareEffect()) {
                    ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software effect present on session");
                    *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
                }
            }
        }
        ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0,
                 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
                 frameCount, mFrameCount);
      } else {
        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
                "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
                "sampleRate=%u mSampleRate=%u "
                "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
                sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
                audio_is_linear_pcm(format),
                channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
        *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
      }
    }
    // For normal PCM streaming tracks, update minimum frame count.
    // 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.
    if (!(*flags & AUDIO_OUTPUT_FLAG_FAST)
            && audio_has_proportional_frames(format) && sharedBuffer == 0) {
        // this must match AudioTrack.cpp calculateMinFrameCount().
        // TODO: Move to a common library
        uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
        uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
        if (minBufCount < 2) {
            minBufCount = 2;
        }
        // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack
        // or the client should compute and pass in a larger buffer request.
        size_t minFrameCount =
                minBufCount * sourceFramesNeededWithTimestretch(
                        sampleRate, mNormalFrameCount,
                        mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/);
        if (frameCount < minFrameCount) { // including frameCount == 0
            frameCount = minFrameCount;
        }
    }
    *pFrameCount = frameCount;

    switch (mType) {

    case DIRECT:
        if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
                        "for output %p with format %#x",
                        sampleRate, format, channelMask, mOutput, mFormat);
                lStatus = BAD_VALUE;
                goto Exit;
            }
        }
        break;

    case OFFLOAD:
        if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
            ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
                    "for output %p with format %#x",
                    sampleRate, format, channelMask, mOutput, mFormat);
            lStatus = BAD_VALUE;
            goto Exit;
        }
        break;

    default:
        if (!audio_is_linear_pcm(format)) {
                ALOGE("createTrack_l() Bad parameter: format %#x \""
                        "for output %p with format %#x",
                        format, mOutput, mFormat);
                lStatus = BAD_VALUE;
                goto Exit;
        }
        if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
            ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
            lStatus = BAD_VALUE;
            goto Exit;
        }
        break;

    }

    lStatus = initCheck();
    if (lStatus != NO_ERROR) {
        ALOGE("createTrack_l() 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->isExternalTrack()) {
                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;
                }
            }
        }

        track = new Track(this, client, streamType, sampleRate, format,
                          channelMask, frameCount, NULL, sharedBuffer,
                          sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);

        lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
        if (lStatus != NO_ERROR) {
            ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
            // track must be cleared from the caller as the caller has the AF lock
            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 & AUDIO_OUTPUT_FLAG_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:
    *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;
    broadcast_l();
}

void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
    Mutex::Autolock _l(mLock);
    mStreamTypes[stream].mute = muted;
    broadcast_l();
}

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;

    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.
        if (track->isExternalTrack()) {
            TrackBase::track_state state = track->mState;
            mLock.unlock();
            status = AudioSystem::startOutput(mId, track->streamType(),
                                              track->sessionId());
            mLock.lock();
            // abort track was stopped/paused while we released the lock
            if (state != track->mState) {
                if (status == NO_ERROR) {
                    mLock.unlock();
                    AudioSystem::stopOutput(mId, track->streamType(),
                                            track->sessionId());
                    mLock.lock();
                }
                return INVALID_OPERATION;
            }
            // abort if start is rejected by audio policy manager
            if (status != NO_ERROR) {
                return PERMISSION_DENIED;
            }
#ifdef ADD_BATTERY_DATA
            // to track the speaker usage
            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
#endif
        }

        // set retry count for buffer fill
        if (track->isOffloaded()) {
            if (track->isStopping_1()) {
                track->mRetryCount = kMaxTrackStopRetriesOffload;
            } else {
                track->mRetryCount = kMaxTrackStartupRetriesOffload;
            }
            track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
        } else {
            track->mRetryCount = kMaxTrackStartupRetries;
            track->mFillingUpStatus =
                    track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
        }

        track->mResetDone = false;
        track->mPresentationCompleteFrames = 0;
        mActiveTracks.add(track);
        mWakeLockUids.add(track->uid());
        mActiveTracksGeneration++;
        mLatestActiveTrack = track;
        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;
    }

    onAddNewTrack_l();
    return status;
}

bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
    track->terminate();
    // active tracks are removed by threadLoop()
    bool trackActive = (mActiveTracks.indexOf(track) >= 0);
    track->mState = TrackBase::STOPPED;
    if (!trackActive) {
        removeTrack_l(track);
    } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
        track->mState = TrackBase::STOPPING_1;
    }

    return trackActive;
}

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::sMaxFastTracks);
        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();
    }
}

void AudioFlinger::PlaybackThread::broadcast_l()
{
    // Thread could be blocked waiting for async
    // so signal it to handle state changes immediately
    // If threadLoop is currently unlocked a signal of mWaitWorkCV will
    // be lost so we also flag to prevent it blocking on mWaitWorkCV
    mSignalPending = true;
    mWaitWorkCV.broadcast();
}

String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
    Mutex::Autolock _l(mLock);
    if (initCheck() != NO_ERROR) {
        return String8();
    }

    char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
    const String8 out_s8(s);
    free(s);
    return out_s8;
}

void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
    sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
    ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);

    desc->mIoHandle = mId;

    switch (event) {
    case AUDIO_OUTPUT_OPENED:
    case AUDIO_OUTPUT_CONFIG_CHANGED:
        desc->mPatch = mPatch;
        desc->mChannelMask = mChannelMask;
        desc->mSamplingRate = mSampleRate;
        desc->mFormat = mFormat;
        desc->mFrameCount = mNormalFrameCount; // FIXME see
                                             // AudioFlinger::frameCount(audio_io_handle_t)
        desc->mFrameCountHAL = mFrameCount;
        desc->mLatency = latency_l();
        break;

    case AUDIO_OUTPUT_CLOSED:
    default:
        break;
    }
    mAudioFlinger->ioConfigChanged(event, desc, pid);
}

void AudioFlinger::PlaybackThread::writeCallback()
{
    ALOG_ASSERT(mCallbackThread != 0);
    mCallbackThread->resetWriteBlocked();
}

void AudioFlinger::PlaybackThread::drainCallback()
{
    ALOG_ASSERT(mCallbackThread != 0);
    mCallbackThread->resetDraining();
}

void AudioFlinger::PlaybackThread::errorCallback()
{
    ALOG_ASSERT(mCallbackThread != 0);
    mCallbackThread->setAsyncError();
}

void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
{
    Mutex::Autolock _l(mLock);
    // reject out of sequence requests
    if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) {
        mWriteAckSequence &= ~1;
        mWaitWorkCV.signal();
    }
}

void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
{
    Mutex::Autolock _l(mLock);
    // reject out of sequence requests
    if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
        mDrainSequence &= ~1;
        mWaitWorkCV.signal();
    }
}

// static
int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
                                                void *param __unused,
                                                void *cookie)
{
    AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
    ALOGV("asyncCallback() event %d", event);
    switch (event) {
    case STREAM_CBK_EVENT_WRITE_READY:
        me->writeCallback();
        break;
    case STREAM_CBK_EVENT_DRAIN_READY:
        me->drainCallback();
        break;
    case STREAM_CBK_EVENT_ERROR:
        me->errorCallback();
        break;
    default:
        ALOGW("asyncCallback() unknown event %d", event);
        break;
    }
    return 0;
}

void AudioFlinger::PlaybackThread::readOutputParameters_l()
{
    // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
    mSampleRate = mOutput->getSampleRate();
    mChannelMask = mOutput->getChannelMask();
    if (!audio_is_output_channel(mChannelMask)) {
        LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
    }
    if ((mType == MIXER || mType == DUPLICATING)
            && !isValidPcmSinkChannelMask(mChannelMask)) {
        LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
                mChannelMask);
    }
    mChannelCount = audio_channel_count_from_out_mask(mChannelMask);

    // Get actual HAL format.
    mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
    // Get format from the shim, which will be different than the HAL format
    // if playing compressed audio over HDMI passthrough.
    mFormat = mOutput->getFormat();
    if (!audio_is_valid_format(mFormat)) {
        LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
    }
    if ((mType == MIXER || mType == DUPLICATING)
            && !isValidPcmSinkFormat(mFormat)) {
        LOG_FATAL("HAL format %#x not supported for mixed output",
                mFormat);
    }
    mFrameSize = mOutput->getFrameSize();
    mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
    mFrameCount = mBufferSize / mFrameSize;
    if (mFrameCount & 15) {
        ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
                mFrameCount);
    }

    if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) &&
            (mOutput->stream->set_callback != NULL)) {
        if (mOutput->stream->set_callback(mOutput->stream,
                                      AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
            mUseAsyncWrite = true;
            mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
        }
    }

    mHwSupportsPause = false;
    if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) {
        if (mOutput->stream->pause != NULL) {
            if (mOutput->stream->resume != NULL) {
                mHwSupportsPause = true;
            } else {
                ALOGW("direct output implements pause but not resume");
            }
        } else if (mOutput->stream->resume != NULL) {
            ALOGW("direct output implements resume but not pause");
        }
    }
    if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) {
        LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume");
    }

    if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
        // For best precision, we use float instead of the associated output
        // device format (typically PCM 16 bit).

        mFormat = AUDIO_FORMAT_PCM_FLOAT;
        mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
        mBufferSize = mFrameSize * mFrameCount;

        // TODO: We currently use the associated output device channel mask and sample rate.
        // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
        // (if a valid mask) to avoid premature downmix.
        // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
        // instead of the output device sample rate to avoid loss of high frequency information.
        // This may need to be updated as MixerThread/OutputTracks are added and not here.
    }

    // Calculate size of normal sink buffer relative to the HAL output buffer size
    double multiplier = 1.0;
    if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
            kUseFastMixer == FastMixer_Dynamic)) {
        size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
        size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * 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 {
            multiplier = floor(multiplier);
        }
    }
    mNormalFrameCount = multiplier * mFrameCount;
    // round up to nearest 16 frames to satisfy AudioMixer
    if (mType == MIXER || mType == DUPLICATING) {
        mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
    }
    ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
            mNormalFrameCount);

    // Check if we want to throttle the processing to no more than 2x normal rate
    mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */);
    mThreadThrottleTimeMs = 0;
    mThreadThrottleEndMs = 0;
    mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate);

    // mSinkBuffer is the sink buffer.  Size is always multiple-of-16 frames.
    // Originally this was int16_t[] array, need to remove legacy implications.
    free(mSinkBuffer);
    mSinkBuffer = NULL;
    // For sink buffer size, we use the frame size from the downstream sink to avoid problems
    // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
    const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
    (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);

    // We resize the mMixerBuffer according to the requirements of the sink buffer which
    // drives the output.
    free(mMixerBuffer);
    mMixerBuffer = NULL;
    if (mMixerBufferEnabled) {
        mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
        mMixerBufferSize = mNormalFrameCount * mChannelCount
                * audio_bytes_per_sample(mMixerBufferFormat);
        (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
    }
    free(mEffectBuffer);
    mEffectBuffer = NULL;
    if (mEffectBufferEnabled) {
        mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only
        mEffectBufferSize = mNormalFrameCount * mChannelCount
                * audio_bytes_per_sample(mEffectBufferFormat);
        (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
    }

    // 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_l() 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(uint32_t *halFrames, uint32_t *dspFrames)
{
    if (halFrames == NULL || dspFrames == NULL) {
        return BAD_VALUE;
    }
    Mutex::Autolock _l(mLock);
    if (initCheck() != NO_ERROR) {
        return INVALID_OPERATION;
    }
    int64_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 = (uint32_t)
                (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0);
        return NO_ERROR;
    } else {
        status_t status;
        uint32_t frames;
        status = mOutput->getRenderPosition(&frames);
        *dspFrames = (size_t)frames;
        return status;
    }
}

// hasAudioSession_l() must be called with ThreadBase::mLock held
uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const
{
    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->isInvalid()) {
            result |= TRACK_SESSION;
            if (track->isFastTrack()) {
                result |= FAST_SESSION;
            }
            break;
        }
    }

    return result;
}

uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t 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->isInvalid()) {
            return AudioSystem::getStrategyForStream(track->streamType());
        }
    }
    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}


AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
{
    Mutex::Autolock _l(mLock);
    return mOutput;
}

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 (count > 0) {
        for (size_t i = 0 ; i < count ; i++) {
            const sp<Track>& track = tracksToRemove.itemAt(i);
            if (track->isExternalTrack()) {
                AudioSystem::stopOutput(mId, track->streamType(),
                                        track->sessionId());
#ifdef ADD_BATTERY_DATA
                // to track the speaker usage
                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
#endif
                if (track->isTerminated()) {
                    AudioSystem::releaseOutput(mId, track->streamType(),
                                               track->sessionId());
                }
            }
        }
    }
}

void AudioFlinger::PlaybackThread::checkSilentMode_l()
{
    if (!mMasterMute) {
        char value[PROPERTY_VALUE_MAX];
        if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
            ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX");
            return;
        }
        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
ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
    mInWrite = true;
    ssize_t bytesWritten;
    const size_t offset = mCurrentWriteLength - mBytesRemaining;

    // If an NBAIO sink is present, use it to write the normal mixer's submix
    if (mNormalSink != 0) {

        const size_t count = mBytesRemaining / mFrameSize;

        ATRACE_BEGIN("write");
        // 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((char *)mSinkBuffer + offset, count);
        ATRACE_END();
        if (framesWritten > 0) {
            bytesWritten = framesWritten * mFrameSize;
        } else {
            bytesWritten = framesWritten;
        }
    // otherwise use the HAL / AudioStreamOut directly
    } else {
        // Direct output and offload threads

        if (mUseAsyncWrite) {
            ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
            mWriteAckSequence += 2;
            mWriteAckSequence |= 1;
            ALOG_ASSERT(mCallbackThread != 0);
            mCallbackThread->setWriteBlocked(mWriteAckSequence);
        }
        // FIXME We should have an implementation of timestamps for direct output threads.
        // They are used e.g for multichannel PCM playback over HDMI.
        bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);

        if (mUseAsyncWrite &&
                ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
            // do not wait for async callback in case of error of full write
            mWriteAckSequence &= ~1;
            ALOG_ASSERT(mCallbackThread != 0);
            mCallbackThread->setWriteBlocked(mWriteAckSequence);
        }
    }

    mNumWrites++;
    mInWrite = false;
    mStandby = false;
    return bytesWritten;
}

void AudioFlinger::PlaybackThread::threadLoop_drain()
{
    if (mOutput->stream->drain) {
        ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
        if (mUseAsyncWrite) {
            ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
            mDrainSequence |= 1;
            ALOG_ASSERT(mCallbackThread != 0);
            mCallbackThread->setDraining(mDrainSequence);
        }
        mOutput->stream->drain(mOutput->stream,
            (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY
                                                : AUDIO_DRAIN_ALL);
    }
}

void AudioFlinger::PlaybackThread::threadLoop_exit()
{
    {
        Mutex::Autolock _l(mLock);
        for (size_t i = 0; i < mTracks.size(); i++) {
            sp<Track> track = mTracks[i];
            track->invalidate();
        }
    }
}

/*
The derived values that are cached:
 - mSinkBufferSize from frame count * frame size
 - mActiveSleepTimeUs from activeSleepTimeUs()
 - mIdleSleepTimeUs from idleSleepTimeUs()
 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least
   kDefaultStandbyTimeInNsecs when connected to an A2DP device.
 - 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()
{
    mSinkBufferSize = mNormalFrameCount * mFrameSize;
    mActiveSleepTimeUs = activeSleepTimeUs();
    mIdleSleepTimeUs = idleSleepTimeUs();

    // make sure standby delay is not too short when connected to an A2DP sink to avoid
    // truncating audio when going to standby.
    mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs;
    if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) {
        if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
            mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
        }
    }
}

bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
{
    ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
            this,  streamType, mTracks.size());
    bool trackMatch = false;
    size_t size = mTracks.size();
    for (size_t i = 0; i < size; i++) {
        sp<Track> t = mTracks[i];
        if (t->streamType() == streamType && t->isExternalTrack()) {
            t->invalidate();
            trackMatch = true;
        }
    }
    return trackMatch;
}

void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
    Mutex::Autolock _l(mLock);
    invalidateTracks_l(streamType);
}

status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
{
    audio_session_t session = chain->sessionId();
    int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
            ? mEffectBuffer : mSinkBuffer);
    bool ownsBuffer = false;

    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
    if (session > AUDIO_SESSION_OUTPUT_MIX) {
        // Only one effect chain can be present in direct output thread and it uses
        // the sink 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->setThread(this);
    chain->setInBuffer(buffer, ownsBuffer);
    chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
            ? mEffectBuffer : mSinkBuffer));
    // 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.
    static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 &&
            AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX,
            "audio_session_t constants misdefined");
    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)
{
    audio_session_t 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(reinterpret_cast<int16_t*>(mSinkBuffer));
                    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;

    mStandbyTimeNs = systemTime();
    nsecs_t lastWriteFinished = -1; // time last server write completed
    int64_t lastFramesWritten = -1; // track changes in timestamp server frames written

    // MIXER
    nsecs_t lastWarning = 0;

    // DUPLICATING
    // FIXME could this be made local to while loop?
    writeFrames = 0;

    int lastGeneration = 0;

    cacheParameters_l();
    mSleepTimeUs = mIdleSleepTimeUs;

    if (mType == MIXER) {
        sleepTimeShift = 0;
    }

    CpuStats cpuStats;
    const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));

    acquireWakeLock();

    // mNBLogWriter->log can only be called while thread mutex mLock is held.
    // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
    // and then that string will be logged at the next convenient opportunity.
    const char *logString = NULL;

    checkSilentMode_l();

    while (!exitPending())
    {
        cpuStats.sample(myName);

        Vector< sp<EffectChain> > effectChains;

        { // scope for mLock

            Mutex::Autolock _l(mLock);

            processConfigEvents_l();

            if (logString != NULL) {
                mNBLogWriter->logTimestamp();
                mNBLogWriter->log(logString);
                logString = NULL;
            }

            // Gather the framesReleased counters for all active tracks,
            // and associate with the sink frames written out.  We need
            // this to convert the sink timestamp to the track timestamp.
            bool kernelLocationUpdate = false;
            if (mNormalSink != 0) {
                // Note: The DuplicatingThread may not have a mNormalSink.
                // We always fetch the timestamp here because often the downstream
                // sink will block while writing.
                ExtendedTimestamp timestamp; // use private copy to fetch
                (void) mNormalSink->getTimestamp(timestamp);

                // We keep track of the last valid kernel position in case we are in underrun
                // and the normal mixer period is the same as the fast mixer period, or there
                // is some error from the HAL.
                if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
                    mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
                            mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
                    mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
                            mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];

                    mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
                            mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
                    mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
                            mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
                }

                if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
                    kernelLocationUpdate = true;
                } else {
                    ALOGVV("getTimestamp error - no valid kernel position");
                }

                // copy over kernel info
                mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
                        timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
                        + mSuspendedFrames; // add frames discarded when suspended
                mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
                        timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
            }
            // mFramesWritten for non-offloaded tracks are contiguous
            // even after standby() is called. This is useful for the track frame
            // to sink frame mapping.
            bool serverLocationUpdate = false;
            if (mFramesWritten != lastFramesWritten) {
                serverLocationUpdate = true;
                lastFramesWritten = mFramesWritten;
            }
            // Only update timestamps if there is a meaningful change.
            // Either the kernel timestamp must be valid or we have written something.
            if (kernelLocationUpdate || serverLocationUpdate) {
                if (serverLocationUpdate) {
                    // use the time before we called the HAL write - it is a bit more accurate
                    // to when the server last read data than the current time here.
                    //
                    // If we haven't written anything, mLastWriteTime will be -1
                    // and we use systemTime().
                    mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
                    mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1
                            ? systemTime() : mLastWriteTime;
                }
                const size_t size = mActiveTracks.size();
                for (size_t i = 0; i < size; ++i) {
                    sp<Track> t = mActiveTracks[i].promote();
                    if (t != 0 && !t->isFastTrack()) {
                        t->updateTrackFrameInfo(
                                t->mAudioTrackServerProxy->framesReleased(),
                                mFramesWritten,
                                mTimestamp);
                    }
                }
            }

            saveOutputTracks();
            if (mSignalPending) {
                // A signal was raised while we were unlocked
                mSignalPending = false;
            } else if (waitingAsyncCallback_l()) {
                if (exitPending()) {
                    break;
                }
                bool released = false;
                if (!keepWakeLock()) {
                    releaseWakeLock_l();
                    released = true;
                }
                mWakeLockUids.clear();
                mActiveTracksGeneration++;
                ALOGV("wait async completion");
                mWaitWorkCV.wait(mLock);
                ALOGV("async completion/wake");
                if (released) {
                    acquireWakeLock_l();
                }
                mStandbyTimeNs = systemTime() + mStandbyDelayNs;
                mSleepTimeUs = 0;

                continue;
            }
            if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
                                   isSuspended()) {
                // put audio hardware into standby after short delay
                if (shouldStandby_l()) {

                    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();
                    mWakeLockUids.clear();
                    mActiveTracksGeneration++;
                    // 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;
                    mBytesRemaining = 0;
                    checkSilentMode_l();

                    mStandbyTimeNs = systemTime() + mStandbyDelayNs;
                    mSleepTimeUs = mIdleSleepTimeUs;
                    if (mType == MIXER) {
                        sleepTimeShift = 0;
                    }

                    continue;
                }
            }
            // mMixerStatusIgnoringFastTracks is also updated internally
            mMixerStatus = prepareTracks_l(&tracksToRemove);

            // compare with previously applied list
            if (lastGeneration != mActiveTracksGeneration) {
                // update wakelock
                updateWakeLockUids_l(mWakeLockUids);
                lastGeneration = mActiveTracksGeneration;
            }

            // 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);
        } // mLock scope ends

        if (mBytesRemaining == 0) {
            mCurrentWriteLength = 0;
            if (mMixerStatus == MIXER_TRACKS_READY) {
                // threadLoop_mix() sets mCurrentWriteLength
                threadLoop_mix();
            } else if ((mMixerStatus != MIXER_DRAIN_TRACK)
                        && (mMixerStatus != MIXER_DRAIN_ALL)) {
                // threadLoop_sleepTime sets mSleepTimeUs to 0 if data
                // must be written to HAL
                threadLoop_sleepTime();
                if (mSleepTimeUs == 0) {
                    mCurrentWriteLength = mSinkBufferSize;
                }
            }
            // Either threadLoop_mix() or threadLoop_sleepTime() should have set
            // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
            // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
            // or mSinkBuffer (if there are no effects).
            //
            // This is done pre-effects computation; if effects change to
            // support higher precision, this needs to move.
            //
            // mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
            // TODO use mSleepTimeUs == 0 as an additional condition.
            if (mMixerBufferValid) {
                void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
                audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;

                // mono blend occurs for mixer threads only (not direct or offloaded)
                // and is handled here if we're going directly to the sink.
                if (requireMonoBlend() && !mEffectBufferValid) {
                    mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount,
                               true /*limit*/);
                }

                memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
                        mNormalFrameCount * mChannelCount);
            }

            mBytesRemaining = mCurrentWriteLength;
            if (isSuspended()) {
                // Simulate write to HAL when suspended (e.g. BT SCO phone call).
                mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer.
                const size_t framesRemaining = mBytesRemaining / mFrameSize;
                mBytesWritten += mBytesRemaining;
                mFramesWritten += framesRemaining;
                mSuspendedFrames += framesRemaining; // to adjust kernel HAL position
                mBytesRemaining = 0;
            }

            // only process effects if we're going to write
            if (mSleepTimeUs == 0 && mType != OFFLOAD) {
                for (size_t i = 0; i < effectChains.size(); i ++) {
                    effectChains[i]->process_l();
                }
            }
        }
        // Process effect chains for offloaded thread even if no audio
        // was read from audio track: process only updates effect state
        // and thus does have to be synchronized with audio writes but may have
        // to be called while waiting for async write callback
        if (mType == OFFLOAD) {
            for (size_t i = 0; i < effectChains.size(); i ++) {
                effectChains[i]->process_l();
            }
        }

        // Only if the Effects buffer is enabled and there is data in the
        // Effects buffer (buffer valid), we need to
        // copy into the sink buffer.
        // TODO use mSleepTimeUs == 0 as an additional condition.
        if (mEffectBufferValid) {
            //ALOGV("writing effect buffer to sink buffer format %#x", mFormat);

            if (requireMonoBlend()) {
                mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount,
                           true /*limit*/);
            }

            memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
                    mNormalFrameCount * mChannelCount);
        }

        // enable changes in effect chain
        unlockEffectChains(effectChains);

        if (!waitingAsyncCallback()) {
            // mSleepTimeUs == 0 means we must write to audio hardware
            if (mSleepTimeUs == 0) {
                ssize_t ret = 0;
                // We save lastWriteFinished here, as previousLastWriteFinished,
                // for throttling. On thread start, previousLastWriteFinished will be
                // set to -1, which properly results in no throttling after the first write.
                nsecs_t previousLastWriteFinished = lastWriteFinished;
                nsecs_t delta = 0;
                if (mBytesRemaining) {
                    // FIXME rewrite to reduce number of system calls
                    mLastWriteTime = systemTime();  // also used for dumpsys
                    ret = threadLoop_write();
                    lastWriteFinished = systemTime();
                    delta = lastWriteFinished - mLastWriteTime;
                    if (ret < 0) {
                        mBytesRemaining = 0;
                    } else {
                        mBytesWritten += ret;
                        mBytesRemaining -= ret;
                        mFramesWritten += ret / mFrameSize;
                    }
                } else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
                        (mMixerStatus == MIXER_DRAIN_ALL)) {
                    threadLoop_drain();
                }
                if (mType == MIXER && !mStandby) {
                    // write blocked detection
                    if (delta > maxPeriod) {
                        mNumDelayedWrites++;
                        if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) {
                            ATRACE_NAME("underrun");
                            ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
                                    (unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
                            lastWarning = lastWriteFinished;
                        }
                    }

                    if (mThreadThrottle
                            && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks)
                            && ret > 0) {                         // we wrote something
                        // Limit MixerThread data processing to no more than twice the
                        // expected processing rate.
                        //
                        // This helps prevent underruns with NuPlayer and other applications
                        // which may set up buffers that are close to the minimum size, or use
                        // deep buffers, and rely on a double-buffering sleep strategy to fill.
                        //
                        // The throttle smooths out sudden large data drains from the device,
                        // e.g. when it comes out of standby, which often causes problems with
                        // (1) mixer threads without a fast mixer (which has its own warm-up)
                        // (2) minimum buffer sized tracks (even if the track is full,
                        //     the app won't fill fast enough to handle the sudden draw).
                        //
                        // Total time spent in last processing cycle equals time spent in
                        // 1. threadLoop_write, as well as time spent in
                        // 2. threadLoop_mix (significant for heavy mixing, especially
                        //                    on low tier processors)

                        // it's OK if deltaMs is an overestimate.
                        const int32_t deltaMs =
                                (lastWriteFinished - previousLastWriteFinished) / 1000000;
                        const int32_t throttleMs = mHalfBufferMs - deltaMs;
                        if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) {
                            usleep(throttleMs * 1000);
                            // notify of throttle start on verbose log
                            ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs,
                                    "mixer(%p) throttle begin:"
                                    " ret(%zd) deltaMs(%d) requires sleep %d ms",
                                    this, ret, deltaMs, throttleMs);
                            mThreadThrottleTimeMs += throttleMs;
                            // Throttle must be attributed to the previous mixer loop's write time
                            // to allow back-to-back throttling.
                            lastWriteFinished += throttleMs * 1000000;
                        } else {
                            uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs;
                            if (diff > 0) {
                                // notify of throttle end on debug log
                                // but prevent spamming for bluetooth
                                ALOGD_IF(!audio_is_a2dp_out_device(outDevice()),
                                        "mixer(%p) throttle end: throttle time(%u)", this, diff);
                                mThreadThrottleEndMs = mThreadThrottleTimeMs;
                            }
                        }
                    }
                }

            } else {
                ATRACE_BEGIN("sleep");
                Mutex::Autolock _l(mLock);
                if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) {
                    mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs));
                }
                ATRACE_END();
            }
        }

        // 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).
    }

    threadLoop_exit();

    if (!mStandby) {
        threadLoop_standby();
        mStandby = true;
    }

    releaseWakeLock();
    mWakeLockUids.clear();
    mActiveTracksGeneration++;

    ALOGV("Thread %p type %d exiting", this, mType);
    return false;
}

// removeTracks_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
{
    size_t count = tracksToRemove.size();
    if (count > 0) {
        for (size_t i=0 ; i<count ; i++) {
            const sp<Track>& track = tracksToRemove.itemAt(i);
            mActiveTracks.remove(track);
            mWakeLockUids.remove(track->uid());
            mActiveTracksGeneration++;
            ALOGV("removeTracks_l removing track on session %d", track->sessionId());
            sp<EffectChain> 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);
            }
        }
    }

}

status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
{
    if (mNormalSink != 0) {
        ExtendedTimestamp ets;
        status_t status = mNormalSink->getTimestamp(ets);
        if (status == NO_ERROR) {
            status = ets.getBestTimestamp(&timestamp);
        }
        return status;
    }
    if ((mType == OFFLOAD || mType == DIRECT)
            && mOutput != NULL && mOutput->stream->get_presentation_position) {
        uint64_t position64;
        int ret = mOutput->getPresentationPosition(&position64, &timestamp.mTime);
        if (ret == 0) {
            timestamp.mPosition = (uint32_t)position64;
            return NO_ERROR;
        }
    }
    return INVALID_OPERATION;
}

status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
                                                          audio_patch_handle_t *handle)
{
    status_t status;
    if (property_get_bool("af.patch_park", false /* default_value */)) {
        // Park FastMixer to avoid potential DOS issues with writing to the HAL
        // or if HAL does not properly lock against access.
        AutoPark<FastMixer> park(mFastMixer);
        status = PlaybackThread::createAudioPatch_l(patch, handle);
    } else {
        status = PlaybackThread::createAudioPatch_l(patch, handle);
    }
    return status;
}

status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
                                                          audio_patch_handle_t *handle)
{
    status_t status = NO_ERROR;

    // store new device and send to effects
    audio_devices_t type = AUDIO_DEVICE_NONE;
    for (unsigned int i = 0; i < patch->num_sinks; i++) {
        type |= patch->sinks[i].ext.device.type;
    }

#ifdef ADD_BATTERY_DATA
    // when changing the audio output device, call addBatteryData to notify
    // the change
    if (mOutDevice != type) {
        uint32_t params = 0;
        // check whether speaker is on
        if (type & 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 (type & deviceWithoutSpeaker) {
            params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
        }

        if (params != 0) {
            addBatteryData(params);
        }
    }
#endif

    for (size_t i = 0; i < mEffectChains.size(); i++) {
        mEffectChains[i]->setDevice_l(type);
    }

    // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when
    // the thread is created so that the first patch creation triggers an ioConfigChanged callback
    bool configChanged = mPrevOutDevice != type;
    mOutDevice = type;
    mPatch = *patch;

    if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
        audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
        status = hwDevice->create_audio_patch(hwDevice,
                                               patch->num_sources,
                                               patch->sources,
                                               patch->num_sinks,
                                               patch->sinks,
                                               handle);
    } else {
        char *address;
        if (strcmp(patch->sinks[0].ext.device.address, "") != 0) {
            //FIXME: we only support address on first sink with HAL version < 3.0
            address = audio_device_address_to_parameter(
                                                        patch->sinks[0].ext.device.type,
                                                        patch->sinks[0].ext.device.address);
        } else {
            address = (char *)calloc(1, 1);
        }
        AudioParameter param = AudioParameter(String8(address));
        free(address);
        param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type);
        status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
                param.toString().string());
        *handle = AUDIO_PATCH_HANDLE_NONE;
    }
    if (configChanged) {
        mPrevOutDevice = type;
        sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
    }
    return status;
}

status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
    status_t status;
    if (property_get_bool("af.patch_park", false /* default_value */)) {
        // Park FastMixer to avoid potential DOS issues with writing to the HAL
        // or if HAL does not properly lock against access.
        AutoPark<FastMixer> park(mFastMixer);
        status = PlaybackThread::releaseAudioPatch_l(handle);
    } else {
        status = PlaybackThread::releaseAudioPatch_l(handle);
    }
    return status;
}

status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
    status_t status = NO_ERROR;

    mOutDevice = AUDIO_DEVICE_NONE;

    if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
        audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
        status = hwDevice->release_audio_patch(hwDevice, handle);
    } else {
        AudioParameter param;
        param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
        status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
                param.toString().string());
    }
    return status;
}

void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
{
    Mutex::Autolock _l(mLock);
    mTracks.add(track);
}

void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
{
    Mutex::Autolock _l(mLock);
    destroyTrack_l(track);
}

void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config)
{
    ThreadBase::getAudioPortConfig(config);
    config->role = AUDIO_PORT_ROLE_SOURCE;
    config->ext.mix.hw_module = mOutput->audioHwDev->handle();
    config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
}

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

AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
        audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
    :   PlaybackThread(audioFlinger, output, id, device, type, systemReady),
        // mAudioMixer below
        // mFastMixer below
        mFastMixerFutex(0),
        mMasterMono(false)
        // mOutputSink below
        // mPipeSink below
        // mNormalSink below
{
    ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, "
            "mFrameCount=%zu, mNormalFrameCount=%zu",
            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
            mNormalFrameCount);
    mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);

    if (type == DUPLICATING) {
        // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
        // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
        // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
        return;
    }
    // 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, mFormat)};
#if !LOG_NDEBUG
    ssize_t index =
#else
    (void)
#endif
            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) {
        audio_format_t fastMixerFormat;
        if (mMixerBufferEnabled && mEffectBufferEnabled) {
            fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
        } else {
            fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
        }
        if (mFormat != fastMixerFormat) {
            // change our Sink format to accept our intermediate precision
            mFormat = fastMixerFormat;
            free(mSinkBuffer);
            mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
            const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
            (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
        }

        // create a MonoPipe to connect our submix to FastMixer
        NBAIO_Format format = mOutputSink->format();
#ifdef TEE_SINK
        NBAIO_Format origformat = format;
#endif
        // adjust format to match that of the Fast Mixer
        ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
        format.mFormat = fastMixerFormat;
        format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;

        // 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;
#if !LOG_NDEBUG || defined(TEE_SINK)
        ssize_t index =
#else
        (void)
#endif
                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
        if (mTeeSinkOutputEnabled) {
            // create a Pipe to archive a copy of FastMixer's output for dumpsys
            Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat);
            const NBAIO_Format offers2[1] = {origformat};
            numCounterOffers = 0;
            index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers);
            ALOG_ASSERT(index == 0);
            mTeeSink = teeSink;
            PipeReader *teeSource = new PipeReader(*teeSink);
            numCounterOffers = 0;
            index = teeSource->negotiate(offers2, 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->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
        fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
        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;
#ifdef TEE_SINK
        state->mTeeSink = mTeeSink.get();
#endif
        mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
        state->mNBLogWriter = mFastMixerNBLogWriter.get();
        sq->end();
        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);

        // start the fast mixer
        mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
        pid_t tid = mFastMixer->getTid();
        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);

#ifdef AUDIO_WATCHDOG
        // create and start the watchdog
        mAudioWatchdog = new AudioWatchdog();
        mAudioWatchdog->setDump(&mAudioWatchdogDump);
        mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
        tid = mAudioWatchdog->getTid();
        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
#endif

    }

    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 != 0) {
        FastMixerStateQueue *sq = mFastMixer->sq();
        FastMixerState *state = sq->begin();
        if (state->mCommand == FastMixerState::COLD_IDLE) {
            int32_t old = android_atomic_inc(&mFastMixerFutex);
            if (old == -1) {
                (void) syscall(__NR_futex, &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*/);
        mFastMixer.clear();
#ifdef AUDIO_WATCHDOG
        if (mAudioWatchdog != 0) {
            mAudioWatchdog->requestExit();
            mAudioWatchdog->requestExitAndWait();
            mAudioWatchdog.clear();
        }
#endif
    }
    mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
    delete mAudioMixer;
}


uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
{
    if (mFastMixer != 0) {
        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);
}

ssize_t 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 != 0) {
        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) {

                // FIXME workaround for first HAL write being CPU bound on some devices
                ATRACE_BEGIN("write");
                mOutput->write((char *)mSinkBuffer, 0);
                ATRACE_END();

                int32_t old = android_atomic_inc(&mFastMixerFutex);
                if (old == -1) {
                    (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
                }
#ifdef AUDIO_WATCHDOG
                if (mAudioWatchdog != 0) {
                    mAudioWatchdog->resume();
                }
#endif
            }
            state->mCommand = FastMixerState::MIX_WRITE;
#ifdef FAST_THREAD_STATISTICS
            mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
                FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
#endif
            sq->end();
            sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
            if (kUseFastMixer == FastMixer_Dynamic) {
                mNormalSink = mPipeSink;
            }
        } else {
            sq->end(false /*didModify*/);
        }
    }
    return PlaybackThread::threadLoop_write();
}

void AudioFlinger::MixerThread::threadLoop_standby()
{
    // Idle the fast mixer if it's currently running
    if (mFastMixer != 0) {
        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();
}

bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
{
    return false;
}

bool AudioFlinger::PlaybackThread::shouldStandby_l()
{
    return !mStandby;
}

bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
{
    Mutex::Autolock _l(mLock);
    return waitingAsyncCallback_l();
}

// 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->standby();
    if (mUseAsyncWrite != 0) {
        // discard any pending drain or write ack by incrementing sequence
        mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
        mDrainSequence = (mDrainSequence + 2) & ~1;
        ALOG_ASSERT(mCallbackThread != 0);
        mCallbackThread->setWriteBlocked(mWriteAckSequence);
        mCallbackThread->setDraining(mDrainSequence);
    }
    mHwPaused = false;
}

void AudioFlinger::PlaybackThread::onAddNewTrack_l()
{
    ALOGV("signal playback thread");
    broadcast_l();
}

void AudioFlinger::PlaybackThread::onAsyncError()
{
    for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
        invalidateTracks((audio_stream_type_t)i);
    }
}

void AudioFlinger::MixerThread::threadLoop_mix()
{
    // mix buffers...
    mAudioMixer->process();
    mCurrentWriteLength = mSinkBufferSize;
    // 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 ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
        sleepTimeShift--;
    }
    mSleepTimeUs = 0;
    mStandbyTimeNs = systemTime() + mStandbyDelayNs;
    //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 (mSleepTimeUs == 0) {
        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
            mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift;
            if (mSleepTimeUs < kMinThreadSleepTimeUs) {
                mSleepTimeUs = 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 {
            mSleepTimeUs = mIdleSleepTimeUs;
        }
    } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
        // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared
        // before effects processing or output.
        if (mMixerBufferValid) {
            memset(mMixerBuffer, 0, mMixerBufferSize);
        } else {
            memset(mSinkBuffer, 0, mSinkBufferSize);
        }
        mSleepTimeUs = 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 != 0) {
        sq = mFastMixer->sq();
        state = sq->begin();
    }

    mMixerBufferValid = false;  // mMixerBuffer has no valid data until appropriate tracks found.
    mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.

    for (size_t i=0 ; i<count ; i++) {
        const 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::sMaxFastTracks);
            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()) &&
                    recentUnderruns > 0) {
                // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
                track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
            } else {
                track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
            }

            // 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->isTerminated()) {
                    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
                    track->disable();
                    // 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::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;
                    int64_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_ALWAYS_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->mChannelMask = track->mChannelMask;
                    fastTrack->mFormat = track->mFormat;
                    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 = 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_ALWAYS_FATAL("fast track %d should have been active; "
                            "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
                            j, track->mState, state->mTrackMask, recentUnderruns,
                            track->sharedBuffer() != 0);
                }
                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
        size_t desiredFrames;
        const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
        AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();

        desiredFrames = sourceFramesNeededWithTimestretch(
                sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
        // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
        // add frames already consumed but not yet released by the resampler
        // because mAudioTrackServerProxy->framesReady() will include these frames
        desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());

        uint32_t minFrames = 1;
        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
            minFrames = desiredFrames;
        }

        size_t framesReady = track->framesReady();
        if (ATRACE_ENABLED()) {
            // I wish we had formatted trace names
            char traceName[16];
            strcpy(traceName, "nRdy");
            int name = track->name();
            if (AudioMixer::TRACK0 <= name &&
                    name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
                name -= AudioMixer::TRACK0;
                traceName[4] = (name / 10) + '0';
                traceName[5] = (name % 10) + '0';
            } else {
                traceName[4] = '?';
                traceName[5] = '?';
            }
            traceName[6] = '\0';
            ATRACE_INT(traceName, framesReady);
        }
        if ((framesReady >= minFrames) && track->isReady() &&
                !track->isPaused() && !track->isTerminated())
        {
            ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this);

            mixedTracks++;

            // track->mainBuffer() != mSinkBuffer or mMixerBuffer means
            // there is an effect chain connected to the track
            chain.clear();
            if (track->mainBuffer() != mSinkBuffer &&
                    track->mainBuffer() != mMixerBuffer) {
                if (mEffectBufferEnabled) {
                    mEffectBufferValid = true; // Later can set directly.
                }
                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);
            // FIXME should not make a decision based on mServer
            } else if (cblk->mServer != 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;       // in U8.24 integer format
            float vlf, vrf, vaf;   // in [0.0, 1.0] float format
            if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
                vl = vr = 0;
                vlf = vrf = vaf = 0.;
                if (track->isPausing()) {
                    track->setPaused();
                }
            } else {

                // read original volumes with volume control
                float typeVolume = mStreamTypes[track->streamType()].volume;
                float v = masterVolume * typeVolume;
                AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
                gain_minifloat_packed_t vlr = proxy->getVolumeLR();
                vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
                vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
                // track volumes come from shared memory, so can't be trusted and must be clamped
                if (vlf > GAIN_FLOAT_UNITY) {
                    ALOGV("Track left volume out of range: %.3g", vlf);
                    vlf = GAIN_FLOAT_UNITY;
                }
                if (vrf > GAIN_FLOAT_UNITY) {
                    ALOGV("Track right volume out of range: %.3g", vrf);
                    vrf = GAIN_FLOAT_UNITY;
                }
                // now apply the master volume and stream type volume
                vlf *= v;
                vrf *= v;
                // assuming master volume and stream type volume each go up to 1.0,
                // then derive vl and vr as U8.24 versions for the effect chain
                const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
                vl = (uint32_t) (scaleto8_24 * vlf);
                vr = (uint32_t) (scaleto8_24 * vrf);
                // vl and vr are now in U8.24 format
                uint16_t sendLevel = proxy->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;
                }
                // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
                vaf = v * sendLevel * (1. / MAX_GAIN_INT);
            }

            // 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;
                // Update remaining floating point volume levels
                vlf = (float)vl / (1 << 24);
                vrf = (float)vr / (1 << 24);
                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;
            }

            // XXX: these things DON'T need to be done each time
            mAudioMixer->setBufferProvider(name, track);
            mAudioMixer->enable(name);

            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
            mAudioMixer->setParameter(
                name,
                AudioMixer::TRACK,
                AudioMixer::FORMAT, (void *)track->format());
            mAudioMixer->setParameter(
                name,
                AudioMixer::TRACK,
                AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
            mAudioMixer->setParameter(
                name,
                AudioMixer::TRACK,
                AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
            // limit track sample rate to 2 x output sample rate, which changes at re-configuration
            uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
            uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
            if (reqSampleRate == 0) {
                reqSampleRate = mSampleRate;
            } else if (reqSampleRate > maxSampleRate) {
                reqSampleRate = maxSampleRate;
            }
            mAudioMixer->setParameter(
                name,
                AudioMixer::RESAMPLE,
                AudioMixer::SAMPLE_RATE,
                (void *)(uintptr_t)reqSampleRate);

            AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
            mAudioMixer->setParameter(
                name,
                AudioMixer::TIMESTRETCH,
                AudioMixer::PLAYBACK_RATE,
                &playbackRate);

            /*
             * Select the appropriate output buffer for the track.
             *
             * Tracks with effects go into their own effects chain buffer
             * and from there into either mEffectBuffer or mSinkBuffer.
             *
             * Other tracks can use mMixerBuffer for higher precision
             * channel accumulation.  If this buffer is enabled
             * (mMixerBufferEnabled true), then selected tracks will accumulate
             * into it.
             *
             */
            if (mMixerBufferEnabled
                    && (track->mainBuffer() == mSinkBuffer
                            || track->mainBuffer() == mMixerBuffer)) {
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,
                        AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,
                        AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
                // TODO: override track->mainBuffer()?
                mMixerBufferValid = true;
            } else {
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,
                        AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
                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 {
            if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
                ALOGV("track(%p) underrun,  framesReady(%zu) < framesDesired(%zd)",
                        track, framesReady, desiredFrames);
                track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
            } else {
                track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
            }

            // 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 s=%08x [NOT READY] on thread %p", name, cblk->mServer, 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;
                int64_t framesWritten = mBytesWritten / mFrameSize;
                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
                    if (track->isStopped()) {
                        track->reset();
                    }
                    tracksToRemove->add(track);
                }
            } else {
                // No buffers for this track. Give it a few chances to
                // fill a buffer, then remove it from active list.
                if (--(track->mRetryCount) <= 0) {
                    ALOGI("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
                    track->disable();
                // 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

    }

    // 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;
        }
    }
    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...
    removeTracks_l(*tracksToRemove);

    if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
        mEffectBufferValid = true;
    }

    if (mEffectBufferValid) {
        // as long as there are effects we should clear the effects buffer, to avoid
        // passing a non-clean buffer to the effect chain
        memset(mEffectBuffer, 0, mEffectBufferSize);
    }
    // sink or mix buffer must be cleared if all tracks are connected to an
    // effect chain as in this case the mixer will not write to the sink or mix buffer
    // and track effects will accumulate into it
    if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
            (mixedTracks == 0 && fastTracks > 0))) {
        // FIXME as a performance optimization, should remember previous zero status
        if (mMixerBufferValid) {
            memset(mMixerBuffer, 0, mMixerBufferSize);
            // TODO: In testing, mSinkBuffer below need not be cleared because
            // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
            // after mixing.
            //
            // To enforce this guarantee:
            // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
            // (mixedTracks == 0 && fastTracks > 0))
            // must imply MIXER_TRACKS_READY.
            // Later, we may clear buffers regardless, and skip much of this logic.
        }
        // FIXME as a performance optimization, should remember previous zero status
        memset