services/audioflinger/AudioFlinger.cpp

/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
**     http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/


#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0

#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>

#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <utils/Atomic.h>

#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <cutils/compiler.h>

#undef ADD_BATTERY_DATA

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

#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>

#include <system/audio.h>
#include <hardware/audio.h>

#include "AudioMixer.h"
#include "AudioFlinger.h"
#include "ServiceUtilities.h"

#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
#include <audio_effects/effect_aec.h>

#include <audio_utils/primitives.h>

#include <powermanager/PowerManager.h>

// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
#ifdef DEBUG_CPU_USAGE
#include <cpustats/CentralTendencyStatistics.h>
#include <cpustats/ThreadCpuUsage.h>
#endif

#include <common_time/cc_helper.h>
#include <common_time/local_clock.h>

#include "FastMixer.h"

// NBAIO implementations
#include "AudioStreamOutSink.h"
#include "MonoPipe.h"
#include "MonoPipeReader.h"
#include "Pipe.h"
#include "PipeReader.h"
#include "SourceAudioBufferProvider.h"

#ifdef HAVE_REQUEST_PRIORITY
#include "SchedulingPolicyService.h"
#endif

#ifdef SOAKER
#include "Soaker.h"
#endif

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

// Note: the following macro is used for extremely verbose logging message.  In
// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
// are so verbose that we want to suppress them even when we have ALOG_ASSERT
// turned on.  Do not uncomment the #def below unless you really know what you
// are doing and want to see all of the extremely verbose messages.
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif

namespace android {

static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
static const char kHardwareLockedString[] = "Hardware lock is taken\n";

static const float MAX_GAIN = 4096.0f;
static const uint32_t MAX_GAIN_INT = 0x1000;

// 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;

static const int kDumpLockRetries = 50;
static const int kDumpLockSleepUs = 20000;

// don't warn about blocked writes or record buffer overflows more often than this
static const nsecs_t kWarningThrottleNs = seconds(5);

// RecordThread loop sleep time upon application overrun or audio HAL read error
static const int kRecordThreadSleepUs = 5000;

// maximum time to wait for setParameters to complete
static const nsecs_t kSetParametersTimeoutNs = seconds(2);

// minimum sleep time for the mixer thread loop when tracks are active but in underrun
static const uint32_t kMinThreadSleepTimeUs = 5000;
// maximum divider applied to the active sleep time in the mixer thread loop
static const uint32_t kMaxThreadSleepTimeShift = 2;

// minimum normal mix buffer size, expressed in milliseconds rather than frames
static const uint32_t kMinNormalMixBufferSizeMs = 20;
// maximum normal mix buffer size
static const uint32_t kMaxNormalMixBufferSizeMs = 24;

nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;

// 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;

static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off"
                              // AudioFlinger::setParameters() updates, other threads read w/o lock

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

#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

static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
    const hw_module_t *mod;
    int rc;

    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
    if (rc) {
        goto out;
    }
    rc = audio_hw_device_open(mod, dev);
    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
    if (rc) {
        goto out;
    }
    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
        rc = BAD_VALUE;
        goto out;
    }
    return 0;

out:
    *dev = NULL;
    return rc;
}

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

AudioFlinger::AudioFlinger()
    : BnAudioFlinger(),
      mPrimaryHardwareDev(NULL),
      mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
      mMasterVolume(1.0f),
      mMasterVolumeSupportLvl(MVS_NONE),
      mMasterMute(false),
      mNextUniqueId(1),
      mMode(AUDIO_MODE_INVALID),
      mBtNrecIsOff(false)
{
}

void AudioFlinger::onFirstRef()
{
    int rc = 0;

    Mutex::Autolock _l(mLock);

    /* TODO: move all this work into an Init() function */
    char val_str[PROPERTY_VALUE_MAX] = { 0 };
    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
        uint32_t int_val;
        if (1 == sscanf(val_str, "%u", &int_val)) {
            mStandbyTimeInNsecs = milliseconds(int_val);
            ALOGI("Using %u mSec as standby time.", int_val);
        } else {
            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
            ALOGI("Using default %u mSec as standby time.",
                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
        }
    }

    mMode = AUDIO_MODE_NORMAL;
    mMasterVolumeSW = 1.0;
    mMasterVolume   = 1.0;
    mHardwareStatus = AUDIO_HW_IDLE;
}

AudioFlinger::~AudioFlinger()
{

    while (!mRecordThreads.isEmpty()) {
        // closeInput() will remove first entry from mRecordThreads
        closeInput(mRecordThreads.keyAt(0));
    }
    while (!mPlaybackThreads.isEmpty()) {
        // closeOutput() will remove first entry from mPlaybackThreads
        closeOutput(mPlaybackThreads.keyAt(0));
    }

    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
        // no mHardwareLock needed, as there are no other references to this
        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
        delete mAudioHwDevs.valueAt(i);
    }
}

static const char * const audio_interfaces[] = {
    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
    AUDIO_HARDWARE_MODULE_ID_A2DP,
    AUDIO_HARDWARE_MODULE_ID_USB,
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))

audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices)
{
    // if module is 0, the request comes from an old policy manager and we should load
    // well known modules
    if (module == 0) {
        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
            loadHwModule_l(audio_interfaces[i]);
        }
    } else {
        // check a match for the requested module handle
        AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module);
        if (audioHwdevice != NULL) {
            return audioHwdevice->hwDevice();
        }
    }
    // then try to find a module supporting the requested device.
    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
        if ((dev->get_supported_devices(dev) & devices) == devices)
            return dev;
    }

    return NULL;
}

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

    result.append("Clients:\n");
    for (size_t i = 0; i < mClients.size(); ++i) {
        sp<Client> client = mClients.valueAt(i).promote();
        if (client != 0) {
            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
            result.append(buffer);
        }
    }

    result.append("Global session refs:\n");
    result.append(" session pid count\n");
    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
        AudioSessionRef *r = mAudioSessionRefs[i];
        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
        result.append(buffer);
    }
    write(fd, result.string(), result.size());
    return NO_ERROR;
}


status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
{
    const size_t SIZE = 256;
    char buffer[SIZE];
    String8 result;
    hardware_call_state hardwareStatus = mHardwareStatus;

    snprintf(buffer, SIZE, "Hardware status: %d\n"
                           "Standby Time mSec: %u\n",
                            hardwareStatus,
                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
    result.append(buffer);
    write(fd, result.string(), result.size());
    return NO_ERROR;
}

status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
{
    const size_t SIZE = 256;
    char buffer[SIZE];
    String8 result;
    snprintf(buffer, SIZE, "Permission Denial: "
            "can't dump AudioFlinger from pid=%d, uid=%d\n",
            IPCThreadState::self()->getCallingPid(),
            IPCThreadState::self()->getCallingUid());
    result.append(buffer);
    write(fd, result.string(), result.size());
    return NO_ERROR;
}

static bool tryLock(Mutex& mutex)
{
    bool locked = false;
    for (int i = 0; i < kDumpLockRetries; ++i) {
        if (mutex.tryLock() == NO_ERROR) {
            locked = true;
            break;
        }
        usleep(kDumpLockSleepUs);
    }
    return locked;
}

status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
    if (!dumpAllowed()) {
        dumpPermissionDenial(fd, args);
    } else {
        // get state of hardware lock
        bool hardwareLocked = tryLock(mHardwareLock);
        if (!hardwareLocked) {
            String8 result(kHardwareLockedString);
            write(fd, result.string(), result.size());
        } else {
            mHardwareLock.unlock();
        }

        bool locked = tryLock(mLock);

        // failed to lock - AudioFlinger is probably deadlocked
        if (!locked) {
            String8 result(kDeadlockedString);
            write(fd, result.string(), result.size());
        }

        dumpClients(fd, args);
        dumpInternals(fd, args);

        // dump playback threads
        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
            mPlaybackThreads.valueAt(i)->dump(fd, args);
        }

        // dump record threads
        for (size_t i = 0; i < mRecordThreads.size(); i++) {
            mRecordThreads.valueAt(i)->dump(fd, args);
        }

        // dump all hardware devs
        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
            dev->dump(dev, fd);
        }
        if (locked) mLock.unlock();
    }
    return NO_ERROR;
}

sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
{
    // If pid is already in the mClients wp<> map, then use that entry
    // (for which promote() is always != 0), otherwise create a new entry and Client.
    sp<Client> client = mClients.valueFor(pid).promote();
    if (client == 0) {
        client = new Client(this, pid);
        mClients.add(pid, client);
    }

    return client;
}

// IAudioFlinger interface


sp<IAudioTrack> AudioFlinger::createTrack(
        pid_t pid,
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        uint32_t channelMask,
        int frameCount,
        IAudioFlinger::track_flags_t flags,
        const sp<IMemory>& sharedBuffer,
        audio_io_handle_t output,
        pid_t tid,
        int *sessionId,
        status_t *status)
{
    sp<PlaybackThread::Track> track;
    sp<TrackHandle> trackHandle;
    sp<Client> client;
    status_t lStatus;
    int lSessionId;

    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
    // but if someone uses binder directly they could bypass that and cause us to crash
    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
        ALOGE("createTrack() invalid stream type %d", streamType);
        lStatus = BAD_VALUE;
        goto Exit;
    }

    {
        Mutex::Autolock _l(mLock);
        PlaybackThread *thread = checkPlaybackThread_l(output);
        PlaybackThread *effectThread = NULL;
        if (thread == NULL) {
            ALOGE("unknown output thread");
            lStatus = BAD_VALUE;
            goto Exit;
        }

        client = registerPid_l(pid);

        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
            // check if an effect chain with the same session ID is present on another
            // output thread and move it here.
            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
                if (mPlaybackThreads.keyAt(i) != output) {
                    uint32_t sessions = t->hasAudioSession(*sessionId);
                    if (sessions & PlaybackThread::EFFECT_SESSION) {
                        effectThread = t.get();
                        break;
                    }
                }
            }
            lSessionId = *sessionId;
        } else {
            // if no audio session id is provided, create one here
            lSessionId = nextUniqueId();
            if (sessionId != NULL) {
                *sessionId = lSessionId;
            }
        }
        ALOGV("createTrack() lSessionId: %d", lSessionId);

        track = thread->createTrack_l(client, streamType, sampleRate, format,
                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);

        // move effect chain to this output thread if an effect on same session was waiting
        // for a track to be created
        if (lStatus == NO_ERROR && effectThread != NULL) {
            Mutex::Autolock _dl(thread->mLock);
            Mutex::Autolock _sl(effectThread->mLock);
            moveEffectChain_l(lSessionId, effectThread, thread, true);
        }

        // Look for sync events awaiting for a session to be used.
        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
                    if (lStatus == NO_ERROR) {
                        track->setSyncEvent(mPendingSyncEvents[i]);
                    } else {
                        mPendingSyncEvents[i]->cancel();
                    }
                    mPendingSyncEvents.removeAt(i);
                    i--;
                }
            }
        }
    }
    if (lStatus == NO_ERROR) {
        trackHandle = new TrackHandle(track);
    } else {
        // remove local strong reference to Client before deleting the Track so that the Client
        // destructor is called by the TrackBase destructor with mLock held
        client.clear();
        track.clear();
    }

Exit:
    if (status != NULL) {
        *status = lStatus;
    }
    return trackHandle;
}

uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
{
    Mutex::Autolock _l(mLock);
    PlaybackThread *thread = checkPlaybackThread_l(output);
    if (thread == NULL) {
        ALOGW("sampleRate() unknown thread %d", output);
        return 0;
    }
    return thread->sampleRate();
}

int AudioFlinger::channelCount(audio_io_handle_t output) const
{
    Mutex::Autolock _l(mLock);
    PlaybackThread *thread = checkPlaybackThread_l(output);
    if (thread == NULL) {
        ALOGW("channelCount() unknown thread %d", output);
        return 0;
    }
    return thread->channelCount();
}

audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
    Mutex::Autolock _l(mLock);
    PlaybackThread *thread = checkPlaybackThread_l(output);
    if (thread == NULL) {
        ALOGW("format() unknown thread %d", output);
        return AUDIO_FORMAT_INVALID;
    }
    return thread->format();
}

size_t AudioFlinger::frameCount(audio_io_handle_t output) const
{
    Mutex::Autolock _l(mLock);
    PlaybackThread *thread = checkPlaybackThread_l(output);
    if (thread == NULL) {
        ALOGW("frameCount() unknown thread %d", output);
        return 0;
    }
    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
    //       should examine all callers and fix them to handle smaller counts
    return thread->frameCount();
}

uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
    Mutex::Autolock _l(mLock);
    PlaybackThread *thread = checkPlaybackThread_l(output);
    if (thread == NULL) {
        ALOGW("latency() unknown thread %d", output);
        return 0;
    }
    return thread->latency();
}

status_t AudioFlinger::setMasterVolume(float value)
{
    status_t ret = initCheck();
    if (ret != NO_ERROR) {
        return ret;
    }

    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    float swmv = value;

    Mutex::Autolock _l(mLock);

    // when hw supports master volume, don't scale in sw mixer
    if (MVS_NONE != mMasterVolumeSupportLvl) {
        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
            AutoMutex lock(mHardwareLock);
            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();

            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
            if (NULL != dev->set_master_volume) {
                dev->set_master_volume(dev, value);
            }
            mHardwareStatus = AUDIO_HW_IDLE;
        }

        swmv = 1.0;
    }

    mMasterVolume   = value;
    mMasterVolumeSW = swmv;
    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
        mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);

    return NO_ERROR;
}

status_t AudioFlinger::setMode(audio_mode_t mode)
{
    status_t ret = initCheck();
    if (ret != NO_ERROR) {
        return ret;
    }

    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }
    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
        ALOGW("Illegal value: setMode(%d)", mode);
        return BAD_VALUE;
    }

    { // scope for the lock
        AutoMutex lock(mHardwareLock);
        mHardwareStatus = AUDIO_HW_SET_MODE;
        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
        mHardwareStatus = AUDIO_HW_IDLE;
    }

    if (NO_ERROR == ret) {
        Mutex::Autolock _l(mLock);
        mMode = mode;
        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
            mPlaybackThreads.valueAt(i)->setMode(mode);
    }

    return ret;
}

status_t AudioFlinger::setMicMute(bool state)
{
    status_t ret = initCheck();
    if (ret != NO_ERROR) {
        return ret;
    }

    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    AutoMutex lock(mHardwareLock);
    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
    mHardwareStatus = AUDIO_HW_IDLE;
    return ret;
}

bool AudioFlinger::getMicMute() const
{
    status_t ret = initCheck();
    if (ret != NO_ERROR) {
        return false;
    }

    bool state = AUDIO_MODE_INVALID;
    AutoMutex lock(mHardwareLock);
    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
    mHardwareStatus = AUDIO_HW_IDLE;
    return state;
}

status_t AudioFlinger::setMasterMute(bool muted)
{
    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    Mutex::Autolock _l(mLock);
    // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
    mMasterMute = muted;
    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
        mPlaybackThreads.valueAt(i)->setMasterMute(muted);

    return NO_ERROR;
}

float AudioFlinger::masterVolume() const
{
    Mutex::Autolock _l(mLock);
    return masterVolume_l();
}

float AudioFlinger::masterVolumeSW() const
{
    Mutex::Autolock _l(mLock);
    return masterVolumeSW_l();
}

bool AudioFlinger::masterMute() const
{
    Mutex::Autolock _l(mLock);
    return masterMute_l();
}

float AudioFlinger::masterVolume_l() const
{
    if (MVS_FULL == mMasterVolumeSupportLvl) {
        float ret_val;
        AutoMutex lock(mHardwareLock);

        mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
        ALOG_ASSERT((NULL != mPrimaryHardwareDev) &&
                    (NULL != mPrimaryHardwareDev->get_master_volume),
                "can't get master volume");

        mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
        mHardwareStatus = AUDIO_HW_IDLE;
        return ret_val;
    }

    return mMasterVolume;
}

status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
        audio_io_handle_t output)
{
    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
        ALOGE("setStreamVolume() invalid stream %d", stream);
        return BAD_VALUE;
    }

    AutoMutex lock(mLock);
    PlaybackThread *thread = NULL;
    if (output) {
        thread = checkPlaybackThread_l(output);
        if (thread == NULL) {
            return BAD_VALUE;
        }
    }

    mStreamTypes[stream].volume = value;

    if (thread == NULL) {
        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
        }
    } else {
        thread->setStreamVolume(stream, value);
    }

    return NO_ERROR;
}

status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
{
    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
        ALOGE("setStreamMute() invalid stream %d", stream);
        return BAD_VALUE;
    }

    AutoMutex lock(mLock);
    mStreamTypes[stream].mute = muted;
    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);

    return NO_ERROR;
}

float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
        return 0.0f;
    }

    AutoMutex lock(mLock);
    float volume;
    if (output) {
        PlaybackThread *thread = checkPlaybackThread_l(output);
        if (thread == NULL) {
            return 0.0f;
        }
        volume = thread->streamVolume(stream);
    } else {
        volume = streamVolume_l(stream);
    }

    return volume;
}

bool AudioFlinger::streamMute(audio_stream_type_t stream) const
{
    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
        return true;
    }

    AutoMutex lock(mLock);
    return streamMute_l(stream);
}

status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    // ioHandle == 0 means the parameters are global to the audio hardware interface
    if (ioHandle == 0) {
        Mutex::Autolock _l(mLock);
        status_t final_result = NO_ERROR;
        {
            AutoMutex lock(mHardwareLock);
            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
                status_t result = dev->set_parameters(dev, keyValuePairs.string());
                final_result = result ?: final_result;
            }
            mHardwareStatus = AUDIO_HW_IDLE;
        }
        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
        AudioParameter param = AudioParameter(keyValuePairs);
        String8 value;
        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
            if (mBtNrecIsOff != btNrecIsOff) {
                for (size_t i = 0; i < mRecordThreads.size(); i++) {
                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
                    RecordThread::RecordTrack *track = thread->track();
                    if (track != NULL) {
                        audio_devices_t device = (audio_devices_t)(
                                thread->device() & AUDIO_DEVICE_IN_ALL);
                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
                        thread->setEffectSuspended(FX_IID_AEC,
                                                   suspend,
                                                   track->sessionId());
                        thread->setEffectSuspended(FX_IID_NS,
                                                   suspend,
                                                   track->sessionId());
                    }
                }
                mBtNrecIsOff = btNrecIsOff;
            }
        }
        String8 screenState;
        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
            bool isOff = screenState == "off";
            if (isOff != (gScreenState & 1)) {
                gScreenState = ((gScreenState & ~1) + 2) | isOff;
            }
        }
        return final_result;
    }

    // hold a strong ref on thread in case closeOutput() or closeInput() is called
    // and the thread is exited once the lock is released
    sp<ThreadBase> thread;
    {
        Mutex::Autolock _l(mLock);
        thread = checkPlaybackThread_l(ioHandle);
        if (thread == NULL) {
            thread = checkRecordThread_l(ioHandle);
        } else if (thread == primaryPlaybackThread_l()) {
            // indicate output device change to all input threads for pre processing
            AudioParameter param = AudioParameter(keyValuePairs);
            int value;
            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
                    (value != 0)) {
                for (size_t i = 0; i < mRecordThreads.size(); i++) {
                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
                }
            }
        }
    }
    if (thread != 0) {
        return thread->setParameters(keyValuePairs);
    }
    return BAD_VALUE;
}

String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
//    ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());

    Mutex::Autolock _l(mLock);

    if (ioHandle == 0) {
        String8 out_s8;

        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
            char *s;
            {
            AutoMutex lock(mHardwareLock);
            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
            s = dev->get_parameters(dev, keys.string());
            mHardwareStatus = AUDIO_HW_IDLE;
            }
            out_s8 += String8(s ? s : "");
            free(s);
        }
        return out_s8;
    }

    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
    if (playbackThread != NULL) {
        return playbackThread->getParameters(keys);
    }
    RecordThread *recordThread = checkRecordThread_l(ioHandle);
    if (recordThread != NULL) {
        return recordThread->getParameters(keys);
    }
    return String8("");
}

size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
    status_t ret = initCheck();
    if (ret != NO_ERROR) {
        return 0;
    }

    AutoMutex lock(mHardwareLock);
    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
    struct audio_config config = {
        sample_rate: sampleRate,
        channel_mask: audio_channel_in_mask_from_count(channelCount),
        format: format,
    };
    size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config);
    mHardwareStatus = AUDIO_HW_IDLE;
    return size;
}

unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
    if (ioHandle == 0) {
        return 0;
    }

    Mutex::Autolock _l(mLock);

    RecordThread *recordThread = checkRecordThread_l(ioHandle);
    if (recordThread != NULL) {
        return recordThread->getInputFramesLost();
    }
    return 0;
}

status_t AudioFlinger::setVoiceVolume(float value)
{
    status_t ret = initCheck();
    if (ret != NO_ERROR) {
        return ret;
    }

    // check calling permissions
    if (!settingsAllowed()) {
        return PERMISSION_DENIED;
    }

    AutoMutex lock(mHardwareLock);
    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
    mHardwareStatus = AUDIO_HW_IDLE;

    return ret;
}

status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
        audio_io_handle_t output) const
{
    status_t status;

    Mutex::Autolock _l(mLock);

    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
    if (playbackThread != NULL) {
        return playbackThread->getRenderPosition(halFrames, dspFrames);
    }

    return BAD_VALUE;
}

void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{

    Mutex::Autolock _l(mLock);

    pid_t pid = IPCThreadState::self()->getCallingPid();
    if (mNotificationClients.indexOfKey(pid) < 0) {
        sp<NotificationClient> notificationClient = new NotificationClient(this,
                                                                            client,
                                                                            pid);
        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);

        mNotificationClients.add(pid, notificationClient);

        sp<IBinder> binder = client->asBinder();
        binder->linkToDeath(notificationClient);

        // the config change is always sent from playback or record threads to avoid deadlock
        // with AudioSystem::gLock
        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
        }

        for (size_t i = 0; i < mRecordThreads.size(); i++) {
            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
        }
    }
}

void AudioFlinger::removeNotificationClient(pid_t pid)
{
    Mutex::Autolock _l(mLock);

    mNotificationClients.removeItem(pid);

    ALOGV("%d died, releasing its sessions", pid);
    size_t num = mAudioSessionRefs.size();
    bool removed = false;
    for (size_t i = 0; i< num; ) {
        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
        ALOGV(" pid %d @ %d", ref->mPid, i);
        if (ref->mPid == pid) {
            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
            mAudioSessionRefs.removeAt(i);
            delete ref;
            removed = true;
            num--;
        } else {
            i++;
        }
    }
    if (removed) {
        purgeStaleEffects_l();
    }
}

// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
{
    size_t size = mNotificationClients.size();
    for (size_t i = 0; i < size; i++) {
        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
                                                                               param2);
    }
}

// removeClient_l() must be called with AudioFlinger::mLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
    mClients.removeItem(pid);
}


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

AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
        uint32_t device, type_t type)
    :   Thread(false),
        mType(type),
        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
        // mChannelMask
        mChannelCount(0),
        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
        mParamStatus(NO_ERROR),
        mStandby(false), mId(id),
        mDevice(device),
        mDeathRecipient(new PMDeathRecipient(this))
{
}

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

void AudioFlinger::ThreadBase::exit()
{
    ALOGV("ThreadBase::exit");
    {
        // 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.signal();
    }
    // When Thread::requestExitAndWait is made virtual and this method is renamed to
    // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
    requestExitAndWait();
}

status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
    status_t status;

    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
    Mutex::Autolock _l(mLock);

    mNewParameters.add(keyValuePairs);
    mWaitWorkCV.signal();
    // wait condition with timeout in case the thread loop has exited
    // before the request could be processed
    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
        status = mParamStatus;
        mWaitWorkCV.signal();
    } else {
        status = TIMED_OUT;
    }
    return status;
}

void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
{
    Mutex::Autolock _l(mLock);
    sendConfigEvent_l(event, param);
}

// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
    ConfigEvent configEvent;
    configEvent.mEvent = event;
    configEvent.mParam = param;
    mConfigEvents.add(configEvent);
    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
    mWaitWorkCV.signal();
}

void AudioFlinger::ThreadBase::processConfigEvents()
{
    mLock.lock();
    while (!mConfigEvents.isEmpty()) {
        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
        ConfigEvent configEvent = mConfigEvents[0];
        mConfigEvents.removeAt(0);
        // release mLock before locking AudioFlinger mLock: lock order is always
        // AudioFlinger then ThreadBase to avoid cross deadlock
        mLock.unlock();
        mAudioFlinger->mLock.lock();
        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
        mAudioFlinger->mLock.unlock();
        mLock.lock();
    }
    mLock.unlock();
}

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

    bool locked = tryLock(mLock);
    if (!locked) {
        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
        write(fd, buffer, strlen(buffer));
    }

    snprintf(buffer, SIZE, "io handle: %d\n", mId);
    result.append(buffer);
    snprintf(buffer, SIZE, "TID: %d\n", getTid());
    result.append(buffer);
    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
    result.append(buffer);
    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
    result.append(buffer);
    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
    result.append(buffer);
    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
    result.append(buffer);
    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
    result.append(buffer);
    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
    result.append(buffer);
    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
    result.append(buffer);
    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
    result.append(buffer);

    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
    result.append(buffer);
    result.append(" Index Command");
    for (size_t i = 0; i < mNewParameters.size(); ++i) {
        snprintf(buffer, SIZE, "\n %02d    ", i);
        result.append(buffer);
        result.append(mNewParameters[i]);
    }

    snprintf(buffer, SIZE, "\n\nPending config events: \n");
    result.append(buffer);
    snprintf(buffer, SIZE, " Index event param\n");
    result.append(buffer);
    for (size_t i = 0; i < mConfigEvents.size(); i++) {
        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
        result.append(buffer);
    }
    result.append("\n");

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

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

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

    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
    write(fd, buffer, strlen(buffer));

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

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

void AudioFlinger::ThreadBase::acquireWakeLock_l()
{
    if (mPowerManager == 0) {
        // use checkService() to avoid blocking if power service is not up yet
        sp<IBinder> binder =
            defaultServiceManager()->checkService(String16("power"));
        if (binder == 0) {
            ALOGW("Thread %s cannot connect to the power manager service", mName);
        } else {
            mPowerManager = interface_cast<IPowerManager>(binder);
            binder->linkToDeath(mDeathRecipient);
        }
    }
    if (mPowerManager != 0) {
        sp<IBinder> binder = new BBinder();
        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
                                                         binder,
                                                         String16(mName));
        if (status == NO_ERROR) {
            mWakeLockToken = binder;
        }
        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
    }
}

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

void AudioFlinger::ThreadBase::releaseWakeLock_l()
{
    if (mWakeLockToken != 0) {
        ALOGV("releaseWakeLock_l() %s", mName);
        if (mPowerManager != 0) {
            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
        }
        mWakeLockToken.clear();
    }
}

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

void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
{
    sp<ThreadBase> thread = mThread.promote();
    if (thread != 0) {
        thread->clearPowerManager();
    }
    ALOGW("power manager service died !!!");
}

void AudioFlinger::ThreadBase::setEffectSuspended(
        const effect_uuid_t *type, bool suspend, int sessionId)
{
    Mutex::Autolock _l(mLock);
    setEffectSuspended_l(type, suspend, sessionId);
}

void AudioFlinger::ThreadBase::setEffectSuspended_l(
        const effect_uuid_t *type, bool suspend, int sessionId)
{
    sp<EffectChain> chain = getEffectChain_l(sessionId);
    if (chain != 0) {
        if (type != NULL) {
            chain->setEffectSuspended_l(type, suspend);
        } else {
            chain->setEffectSuspendedAll_l(suspend);
        }
    }

    updateSuspendedSessions_l(type, suspend, sessionId);
}

void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
{
    ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
    if (index < 0) {
        return;
    }

    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
            mSuspendedSessions.editValueAt(index);

    for (size_t i = 0; i < sessionEffects.size(); i++) {
        sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
        for (int j = 0; j < desc->mRefCount; j++) {
            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
                chain->setEffectSuspendedAll_l(true);
            } else {
                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
                    desc->mType.timeLow);
                chain->setEffectSuspended_l(&desc->mType, true);
            }
        }
    }
}

void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
                                                         bool suspend,
                                                         int sessionId)
{
    ssize_t index = mSuspendedSessions.indexOfKey(sessionId);

    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;

    if (suspend) {
        if (index >= 0) {
            sessionEffects = mSuspendedSessions.editValueAt(index);
        } else {
            mSuspendedSessions.add(sessionId, sessionEffects);
        }
    } else {
        if (index < 0) {
            return;
        }
        sessionEffects = mSuspendedSessions.editValueAt(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) {
                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
            }
            sessionEffects.add(key, desc);
            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
        }
        desc->mRefCount++;
    } else {
        if (index < 0) {
            return;
        }
        desc = sessionEffects.valueAt(index);
        if (--desc->mRefCount == 0) {
            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
            sessionEffects.removeItemsAt(index);
            if (sessionEffects.isEmpty()) {
                ALOGV("updateSuspendedSessions_l() restore removing session %d",
                                 sessionId);
                mSuspendedSessions.removeItem(sessionId);
            }
        }
    }
    if (!sessionEffects.isEmpty()) {
        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
    }
}

void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
                                                            bool enabled,
                                                            int sessionId)
{
    Mutex::Autolock _l(mLock);
    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
}

void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
                                                            bool enabled,
                                                            int sessionId)
{
    if (mType != RECORD) {
        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
        // another session. This gives the priority to well behaved effect control panels
        // and applications not using global effects.
        // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
        // global effects
        if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
        }
    }

    sp<EffectChain> chain = getEffectChain_l(sessionId);
    if (chain != 0) {
        chain->checkSuspendOnEffectEnabled(effect, enabled);
    }
}

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

AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
                                             AudioStreamOut* output,
                                             audio_io_handle_t id,
                                             uint32_t device,
                                             type_t type)
    :   ThreadBase(audioFlinger, id, device, type),
        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
        // Assumes constructor is called by AudioFlinger with it's mLock held,
        // but it would be safer to explicitly pass initial masterMute as parameter
        mMasterMute(audioFlinger->masterMute_l()),
        // mStreamTypes[] initialized in constructor body
        mOutput(output),
        // Assumes constructor is called by AudioFlinger with it's mLock held,
        // but it would be safer to explicitly pass initial masterVolume as parameter
        mMasterVolume(audioFlinger->masterVolumeSW_l()),
        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
        mMixerStatus(MIXER_IDLE),
        mMixerStatusIgnoringFastTracks(MIXER_IDLE),
        standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
        mScreenState(gScreenState),
        // index 0 is reserved for normal mixer's submix
        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
{
    snprintf(mName, kNameLength, "AudioOut_%X", id);

    readOutputParameters();

    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
            stream = (audio_stream_type_t) (stream + 1)) {
        mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
        mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
    }
    // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
    // because mAudioFlinger doesn't have one to copy from
}

AudioFlinger::PlaybackThread::~PlaybackThread()
{
    delete [] mMixBuffer;
}

status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
    dumpInternals(fd, args);
    dumpTracks(fd, args);
    dumpEffectChains(fd, args);
    return NO_ERROR;
}

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

    result.appendFormat("Output thread %p stream volumes in dB:\n    ", this);
    for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
        const stream_type_t *st = &mStreamTypes[i];
        if (i > 0) {
            result.appendFormat(", ");
        }
        result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
        if (st->mute) {
            result.append("M");
        }
    }
    result.append("\n");
    write(fd, result.string(), result.length());
    result.clear();

    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
    result.append(buffer);
    Track::appendDumpHeader(result);
    for (size_t i = 0; i < mTracks.size(); ++i) {
        sp<Track> track = mTracks[i];
        if (track != 0) {
            track->dump(buffer, SIZE);
            result.append(buffer);
        }
    }

    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
    result.append(buffer);
    Track::appendDumpHeader(result);
    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
        sp<Track> track = mActiveTracks[i].promote();
        if (track != 0) {
            track->dump(buffer, SIZE);
            result.append(buffer);
        }
    }
    write(fd, result.string(), result.size());

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

    return NO_ERROR;
}

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

    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
    result.append(buffer);
    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
    result.append(buffer);
    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
    result.append(buffer);
    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
    result.append(buffer);
    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
    result.append(buffer);
    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
    result.append(buffer);
    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
    result.append(buffer);
    write(fd, result.string(), result.size());
    fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);

    dumpBase(fd, args);

    return NO_ERROR;
}

// Thread virtuals
status_t AudioFlinger::PlaybackThread::readyToRun()
{
    status_t status = initCheck();
    if (status == NO_ERROR) {
        ALOGI("AudioFlinger's thread %p ready to run", this);
    } else {
        ALOGE("No working audio driver found.");
    }
    return status;
}

void AudioFlinger::PlaybackThread::onFirstRef()
{
    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
}

// 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,
        uint32_t channelMask,
        int frameCount,
        const sp<IMemory>& sharedBuffer,
        int sessionId,
        IAudioFlinger::track_flags_t flags,
        pid_t tid,
        status_t *status)
{
    sp<Track> track;
    status_t lStatus;

    bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0;

    // client expresses a preference for FAST, but we get the final say
    if (flags & IAudioFlinger::TRACK_FAST) {
      if (
            // not timed
            (!isTimed) &&
            // either of these use cases:
            (
              // use case 1: shared buffer with any frame count
              (
                (sharedBuffer != 0)
              ) ||
              // use case 2: callback handler and frame count is default or at least as large as HAL
              (
                (tid != -1) &&
                ((frameCount == 0) ||
                (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below
              )
            ) &&
            // PCM data
            audio_is_linear_pcm(format) &&
            // mono or stereo
            ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
              (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
            // hardware sample rate
            (sampleRate == mSampleRate) &&
#endif
            // normal mixer has an associated fast mixer
            hasFastMixer() &&
            // there are sufficient fast track slots available
            (mFastTrackAvailMask != 0)
            // FIXME test that MixerThread for this fast track has a capable output HAL
            // FIXME add a permission test also?
        ) {
        // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
        if (frameCount == 0) {
            frameCount = mFrameCount * 2;   // FIXME * 2 is due to SRC jitter, should be computed
        }
        ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
                frameCount, mFrameCount);
      } else {
        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
                "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d "
                "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
                isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
                audio_is_linear_pcm(format),
                channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
        flags &= ~IAudioFlinger::TRACK_FAST;
        // For compatibility with AudioTrack calculation, buffer depth is forced
        // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
        // This is probably too conservative, but legacy application code may depend on it.
        // If you change this calculation, also review the start threshold which is related.
        uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
        uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
        if (minBufCount < 2) {
            minBufCount = 2;
        }
        int minFrameCount = mNormalFrameCount * minBufCount;
        if (frameCount < minFrameCount) {
            frameCount = minFrameCount;
        }
      }
    }

    if (mType == DIRECT) {
        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
                        "for output %p with format %d",
                        sampleRate, format, channelMask, mOutput, mFormat);
                lStatus = BAD_VALUE;
                goto Exit;
            }
        }
    } else {
        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
        if (sampleRate > mSampleRate*2) {
            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
            lStatus = BAD_VALUE;
            goto Exit;
        }
    }

    lStatus = initCheck();
    if (lStatus != NO_ERROR) {
        ALOGE("Audio driver not initialized.");
        goto Exit;
    }

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

        // all tracks in same audio session must share the same routing strategy otherwise
        // conflicts will happen when tracks are moved from one output to another by audio policy
        // manager
        uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
        for (size_t i = 0; i < mTracks.size(); ++i) {
            sp<Track> t = mTracks[i];
            if (t != 0 && !t->isOutputTrack()) {
                uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
                if (sessionId == t->sessionId() && strategy != actual) {
                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
                            strategy, actual);
                    lStatus = BAD_VALUE;
                    goto Exit;
                }
            }
        }

        if (!isTimed) {
            track = new Track(this, client, streamType, sampleRate, format,
                    channelMask, frameCount, sharedBuffer, sessionId, flags);
        } else {
            track = TimedTrack::create(this, client, streamType, sampleRate, format,
                    channelMask, frameCount, sharedBuffer, sessionId);
        }
        if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
            lStatus = NO_MEMORY;
            goto Exit;
        }
        mTracks.add(track);

        sp<EffectChain> chain = getEffectChain_l(sessionId);
        if (chain != 0) {
            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
            track->setMainBuffer(chain->inBuffer());
            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
            chain->incTrackCnt();
        }
    }

#ifdef HAVE_REQUEST_PRIORITY
    if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
        pid_t callingPid = IPCThreadState::self()->getCallingPid();
        // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
        // so ask activity manager to do this on our behalf
        int err = requestPriority(callingPid, tid, 1);
        if (err != 0) {
            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                    1, callingPid, tid, err);
        }
    }
#endif

    lStatus = NO_ERROR;

Exit:
    if (status) {
        *status = lStatus;
    }
    return track;
}

uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const
{
    if (mFastMixer != NULL) {
        MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
        latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
    }
    return latency;
}

uint32_t AudioFlinger::PlaybackThread::correctLatency(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(mOutput->stream->get_latency(mOutput->stream));
    } else {
        return 0;
    }
}

void AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
    Mutex::Autolock _l(mLock);
    mMasterVolume = value;
}

void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
    Mutex::Autolock _l(mLock);
    setMasterMute_l(muted);
}

void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
    Mutex::Autolock _l(mLock);
    mStreamTypes[stream].volume = value;
}

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

float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
{
    Mutex::Autolock _l(mLock);
    return mStreamTypes[stream].volume;
}

// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
    status_t status = ALREADY_EXISTS;

    // set retry count for buffer fill
    track->mRetryCount = kMaxTrackStartupRetries;
    if (mActiveTracks.indexOf(track) < 0) {
        // the track is newly added, make sure it fills up all its
        // buffers before playing. This is to ensure the client will
        // effectively get the latency it requested.
        track->mFillingUpStatus = Track::FS_FILLING;
        track->mResetDone = false;
        track->mPresentationCompleteFrames = 0;
        mActiveTracks.add(track);
        if (track->mainBuffer() != mMixBuffer) {
            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
            if (chain != 0) {
                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
                chain->incActiveTrackCnt();
            }
        }

        status = NO_ERROR;
    }

    ALOGV("mWaitWorkCV.broadcast");
    mWaitWorkCV.broadcast();

    return status;
}

// destroyTrack_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
    track->mState = TrackBase::TERMINATED;
    // active tracks are removed by threadLoop()
    if (mActiveTracks.indexOf(track) < 0) {
        removeTrack_l(track);
    }
}

void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
{
    track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
    mTracks.remove(track);
    deleteTrackName_l(track->name());
    // redundant as track is about to be destroyed, for dumpsys only
    track->mName = -1;
    if (track->isFastTrack()) {
        int index = track->mFastIndex;
        ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
        ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
        mFastTrackAvailMask |= 1 << index;
        // redundant as track is about to be destroyed, for dumpsys only
        track->mFastIndex = -1;
    }
    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
    if (chain != 0) {
        chain->decTrackCnt();
    }
}

String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
    String8 out_s8 = String8("");
    char *s;

    Mutex::Autolock _l(mLock);
    if (initCheck() != NO_ERROR) {
        return out_s8;
    }

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

// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
    AudioSystem::OutputDescriptor desc;
    void *param2 = NULL;

    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);

    switch (event) {
    case AudioSystem::OUTPUT_OPENED:
    case AudioSystem::OUTPUT_CONFIG_CHANGED:
        desc.channels = mChannelMask;
        desc.samplingRate = mSampleRate;
        desc.format = mFormat;
        desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t)
        desc.latency = latency();
        param2 = &desc;
        break;

    case AudioSystem::STREAM_CONFIG_CHANGED:
        param2 = &param;
    case AudioSystem::OUTPUT_CLOSED:
    default:
        break;
    }
    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}

void AudioFlinger::PlaybackThread::readOutputParameters()
{
    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
    mChannelCount = (uint16_t)popcount(mChannelMask);
    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
    if (mFrameCount & 15) {
        ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
                mFrameCount);
    }

    // Calculate size of normal mix buffer relative to the HAL output buffer size
    double multiplier = 1.0;
    if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) {
        size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
        size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
        // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
        minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
        maxNormalFrameCount = maxNormalFrameCount & ~15;
        if (maxNormalFrameCount < minNormalFrameCount) {
            maxNormalFrameCount = minNormalFrameCount;
        }
        multiplier = (double) minNormalFrameCount / (double) mFrameCount;
        if (multiplier <= 1.0) {
            multiplier = 1.0;
        } else if (multiplier <= 2.0) {
            if (2 * mFrameCount <= maxNormalFrameCount) {
                multiplier = 2.0;
            } else {
                multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
            }
        } else {
            // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC
            // (it would be unusual for the normal mix buffer size to not be a multiple of fast
            // track, but we sometimes have to do this to satisfy the maximum frame count constraint)
            // FIXME this rounding up should not be done if no HAL SRC
            uint32_t truncMult = (uint32_t) multiplier;
            if ((truncMult & 1)) {
                if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
                    ++truncMult;
                }
            }
            multiplier = (double) truncMult;
        }
    }
    mNormalFrameCount = multiplier * mFrameCount;
    // round up to nearest 16 frames to satisfy AudioMixer
    mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount);

    delete[] mMixBuffer;
    mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
    memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));

    // force reconfiguration of effect chains and engines to take new buffer size and audio
    // parameters into account
    // Note that mLock is not held when readOutputParameters() is called from the constructor
    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
    // matter.
    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
    Vector< sp<EffectChain> > effectChains = mEffectChains;
    for (size_t i = 0; i < effectChains.size(); i ++) {
        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
    }
}


status_t AudioFlinger::PlaybackThread::getRenderPosition(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;
    }
    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);

    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
}

uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
{
    Mutex::Autolock _l(mLock);
    uint32_t result = 0;
    if (getEffectChain_l(sessionId) != 0) {
        result = EFFECT_SESSION;
    }

    for (size_t i = 0; i < mTracks.size(); ++i) {
        sp<Track> track = mTracks[i];
        if (sessionId == track->sessionId() &&
                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
            result |= TRACK_SESSION;
            break;
        }
    }

    return result;
}

uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
{
    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
    }
    for (size_t i = 0; i < mTracks.size(); i++) {
        sp<Track> track = mTracks[i];
        if (sessionId == track->sessionId() &&
                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
            return AudioSystem::getStrategyForStream(track->streamType());
        }
    }
    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}


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

AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
{
    Mutex::Autolock _l(mLock);
    AudioStreamOut *output = mOutput;
    mOutput = NULL;
    // FIXME FastMixer might also have a raw ptr to mOutputSink;
    //       must push a NULL and wait for ack
    mOutputSink.clear();
    mPipeSink.clear();
    mNormalSink.clear();
    return output;
}

// this method must always be called either with ThreadBase mLock held or inside the thread loop
audio_stream_t* AudioFlinger::PlaybackThread::stream() const
{
    if (mOutput == NULL) {
        return NULL;
    }
    return &mOutput->stream->common;
}

uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
{
    return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
}

status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
{
    if (!isValidSyncEvent(event)) {
        return BAD_VALUE;
    }

    Mutex::Autolock _l(mLock);

    for (size_t i = 0; i < mTracks.size(); ++i) {
        sp<Track> track = mTracks[i];
        if (event->triggerSession() == track->sessionId()) {
            track->setSyncEvent(event);
            return NO_ERROR;
        }
    }

    return NAME_NOT_FOUND;
}

bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event)
{
    switch (event->type()) {
    case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE:
        return true;
    default:
        break;
    }
    return false;
}

void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
{
    size_t count = tracksToRemove.size();
    if (CC_UNLIKELY(count)) {
        for (size_t i = 0 ; i < count ; i++) {
            const sp<Track>& track = tracksToRemove.itemAt(i);
            if ((track->sharedBuffer() != 0) &&
                    (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) {
                AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
            }
        }
    }

}

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

AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
        audio_io_handle_t id, uint32_t device, type_t type)
    :   PlaybackThread(audioFlinger, output, id, device, type),
        // mAudioMixer below
#ifdef SOAKER
        mSoaker(NULL),
#endif
        // mFastMixer below
        mFastMixerFutex(0)
        // mOutputSink below
        // mPipeSink below
        // mNormalSink below
{
    ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type);
    ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
            "mFrameCount=%d, mNormalFrameCount=%d",
            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
            mNormalFrameCount);
    mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);

    // FIXME - Current mixer implementation only supports stereo output
    if (mChannelCount == 1) {
        ALOGE("Invalid audio hardware channel count");
    }

    // create an NBAIO sink for the HAL output stream, and negotiate
    mOutputSink = new AudioStreamOutSink(output->stream);
    size_t numCounterOffers = 0;
    const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
    ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
    ALOG_ASSERT(index == 0);

    // initialize fast mixer depending on configuration
    bool initFastMixer;
    switch (kUseFastMixer) {
    case FastMixer_Never:
        initFastMixer = false;
        break;
    case FastMixer_Always:
        initFastMixer = true;
        break;
    case FastMixer_Static:
    case FastMixer_Dynamic:
        initFastMixer = mFrameCount < mNormalFrameCount;
        break;
    }
    if (initFastMixer) {

        // create a MonoPipe to connect our submix to FastMixer
        NBAIO_Format format = mOutputSink->format();
        // This pipe depth compensates for scheduling latency of the normal mixer thread.
        // When it wakes up after a maximum latency, it runs a few cycles quickly before
        // finally blocking.  Note the pipe implementation rounds up the request to a power of 2.
        MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
        const NBAIO_Format offers[1] = {format};
        size_t numCounterOffers = 0;
        ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
        ALOG_ASSERT(index == 0);
        monoPipe->setAvgFrames((mScreenState & 1) ?
                (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
        mPipeSink = monoPipe;

#ifdef TEE_SINK_FRAMES
        // create a Pipe to archive a copy of FastMixer's output for dumpsys
        Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format);
        numCounterOffers = 0;
        index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
        ALOG_ASSERT(index == 0);
        mTeeSink = teeSink;
        PipeReader *teeSource = new PipeReader(*teeSink);
        numCounterOffers = 0;
        index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
        ALOG_ASSERT(index == 0);
        mTeeSource = teeSource;
#endif

#ifdef SOAKER
        // create a soaker as workaround for governor issues
        mSoaker = new Soaker();
        // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE
        mSoaker->run("Soaker", PRIORITY_LOWEST);
#endif

        // create fast mixer and configure it initially with just one fast track for our submix
        mFastMixer = new FastMixer();
        FastMixerStateQueue *sq = mFastMixer->sq();
#ifdef STATE_QUEUE_DUMP
        sq->setObserverDump(&mStateQueueObserverDump);
        sq->setMutatorDump(&mStateQueueMutatorDump);
#endif
        FastMixerState *state = sq->begin();
        FastTrack *fastTrack = &state->mFastTracks[0];
        // wrap the source side of the MonoPipe to make it an AudioBufferProvider
        fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
        fastTrack->mVolumeProvider = NULL;
        fastTrack->mGeneration++;
        state->mFastTracksGen++;
        state->mTrackMask = 1;
        // fast mixer will use the HAL output sink
        state->mOutputSink = mOutputSink.get();
        state->mOutputSinkGen++;
        state->mFrameCount = mFrameCount;
        state->mCommand = FastMixerState::COLD_IDLE;
        // already done in constructor initialization list
        //mFastMixerFutex = 0;
        state->mColdFutexAddr = &mFastMixerFutex;
        state->mColdGen++;
        state->mDumpState = &mFastMixerDumpState;
        state->mTeeSink = mTeeSink.get();
        sq->end();
        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);

        // start the fast mixer
        mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
#ifdef HAVE_REQUEST_PRIORITY
        pid_t tid = mFastMixer->getTid();
        int err = requestPriority(getpid_cached, tid, 2);
        if (err != 0) {
            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                    2, getpid_cached, tid, err);
        }
#endif

#ifdef AUDIO_WATCHDOG
        // create and start the watchdog
        mAudioWatchdog = new AudioWatchdog();
        mAudioWatchdog->setDump(&mAudioWatchdogDump);
        mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
        tid = mAudioWatchdog->getTid();
        err = requestPriority(getpid_cached, tid, 1);
        if (err != 0) {
            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                    1, getpid_cached, tid, err);
        }
#endif

    } else {
        mFastMixer = NULL;
    }

    switch (kUseFastMixer) {
    case FastMixer_Never:
    case FastMixer_Dynamic:
        mNormalSink = mOutputSink;
        break;
    case FastMixer_Always:
        mNormalSink = mPipeSink;
        break;
    case FastMixer_Static:
        mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
        break;
    }
}

AudioFlinger::MixerThread::~MixerThread()
{
    if (mFastMixer != NULL) {
        FastMixerStateQueue *sq = mFastMixer->sq();
        FastMixerState *state = sq->begin();
        if (state->mCommand == FastMixerState::COLD_IDLE) {
            int32_t old = android_atomic_inc(&mFastMixerFutex);
            if (old == -1) {
                __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
            }
        }
        state->mCommand = FastMixerState::EXIT;
        sq->end();
        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
        mFastMixer->join();
        // Though the fast mixer thread has exited, it's state queue is still valid.
        // We'll use that extract the final state which contains one remaining fast track
        // corresponding to our sub-mix.
        state = sq->begin();
        ALOG_ASSERT(state->mTrackMask == 1);
        FastTrack *fastTrack = &state->mFastTracks[0];
        ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
        delete fastTrack->mBufferProvider;
        sq->end(false /*didModify*/);
        delete mFastMixer;
#ifdef SOAKER
        if (mSoaker != NULL) {
            mSoaker->requestExitAndWait();
        }
        delete mSoaker;
#endif
        if (mAudioWatchdog != 0) {
            mAudioWatchdog->requestExit();
            mAudioWatchdog->requestExitAndWait();
            mAudioWatchdog.clear();
        }
    }
    delete mAudioMixer;
}

class CpuStats {
public:
    CpuStats();
    void sample(const String8 &title);
#ifdef DEBUG_CPU_USAGE
private:
    ThreadCpuUsage mCpuUsage;           // instantaneous thread CPU usage in wall clock ns
    CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns

    CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles

    int mCpuNum;                        // thread's current CPU number
    int mCpukHz;                        // frequency of thread's current CPU in kHz
#endif
};

CpuStats::CpuStats()
#ifdef DEBUG_CPU_USAGE
    : mCpuNum(-1), mCpukHz(-1)
#endif
{
}

void CpuStats::sample(const String8 &title) {
#ifdef DEBUG_CPU_USAGE
    // get current thread's delta CPU time in wall clock ns
    double wcNs;
    bool valid = mCpuUsage.sampleAndEnable(wcNs);

    // record sample for wall clock statistics
    if (valid) {
        mWcStats.sample(wcNs);
    }

    // get the current CPU number
    int cpuNum = sched_getcpu();

    // get the current CPU frequency in kHz
    int cpukHz = mCpuUsage.getCpukHz(cpuNum);

    // check if either CPU number or frequency changed
    if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
        mCpuNum = cpuNum;
        mCpukHz = cpukHz;
        // ignore sample for purposes of cycles
        valid = false;
    }

    // if no change in CPU number or frequency, then record sample for cycle statistics
    if (valid && mCpukHz > 0) {
        double cycles = wcNs * cpukHz * 0.000001;
        mHzStats.sample(cycles);
    }

    unsigned n = mWcStats.n();
    // mCpuUsage.elapsed() is expensive, so don't call it every loop
    if ((n & 127) == 1) {
        long long elapsed = mCpuUsage.elapsed();
        if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
            double perLoop = elapsed / (double) n;
            double perLoop100 = perLoop * 0.01;
            double perLoop1k = perLoop * 0.001;
            double mean = mWcStats.mean();
            double stddev = mWcStats.stddev();
            double minimum = mWcStats.minimum();
            double maximum = mWcStats.maximum();
            double meanCycles = mHzStats.mean();
            double stddevCycles = mHzStats.stddev();
            double minCycles = mHzStats.minimum();
            double maxCycles = mHzStats.maximum();
            mCpuUsage.resetElapsed();
            mWcStats.reset();
            mHzStats.reset();
            ALOGD("CPU usage for %s over past %.1f secs\n"
                "  (%u mixer loops at %.1f mean ms per loop):\n"
                "  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
                "  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
                "  MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
                    title.string(),
                    elapsed * .000000001, n, perLoop * .000001,
                    mean * .001,
                    stddev * .001,
                    minimum * .001,
                    maximum * .001,
                    mean / perLoop100,
                    stddev / perLoop100,
                    minimum / perLoop100,
                    maximum / perLoop100,
                    meanCycles / perLoop1k,
                    stddevCycles / perLoop1k,
                    minCycles / perLoop1k,
                    maxCycles / perLoop1k);

        }
    }
#endif
};

void AudioFlinger::PlaybackThread::checkSilentMode_l()
{
    if (!mMasterMute) {
        char value[PROPERTY_VALUE_MAX];
        if (property_get("ro.audio.silent", value, "0") > 0) {
            char *endptr;
            unsigned long ul = strtoul(value, &endptr, 0);
            if (*endptr == '\0' && ul != 0) {
                ALOGD("Silence is golden");
                // The setprop command will not allow a property to be changed after
                // the first time it is set, so we don't have to worry about un-muting.
                setMasterMute_l(true);
            }
        }
    }
}

bool AudioFlinger::PlaybackThread::threadLoop()
{
    Vector< sp<Track> > tracksToRemove;

    standbyTime = systemTime();

    // MIXER
    nsecs_t lastWarning = 0;
if (mType == MIXER) {
    longStandbyExit = false;
}

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

    cacheParameters_l();
    sleepTime = idleSleepTime;

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

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

    acquireWakeLock();

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

        Vector< sp<EffectChain> > effectChains;

        processConfigEvents();

        { // scope for mLock

            Mutex::Autolock _l(mLock);

            if (checkForNewParameters_l()) {
                cacheParameters_l();
            }

            saveOutputTracks();

            // put audio hardware into standby after short delay
            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
                        mSuspended > 0)) {
                if (!mStandby) {

                    threadLoop_standby();

                    mStandby = true;
                    mBytesWritten = 0;
                }

                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
                    // we're about to wait, flush the binder command buffer
                    IPCThreadState::self()->flushCommands();

                    clearOutputTracks();

                    if (exitPending()) break;

                    releaseWakeLock_l();
                    // wait until we have something to do...
                    ALOGV("%s going to sleep", myName.string());
                    mWaitWorkCV.wait(mLock);
                    ALOGV("%s waking up", myName.string());
                    acquireWakeLock_l();

                    mMixerStatus = MIXER_IDLE;
                    mMixerStatusIgnoringFastTracks = MIXER_IDLE;

                    checkSilentMode_l();

                    standbyTime = systemTime() + standbyDelay;
                    sleepTime = idleSleepTime;
                    if (mType == MIXER) {
                        sleepTimeShift = 0;
                    }

                    continue;
                }
            }

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

            // prevent any changes in effect chain list and in each effect chain
            // during mixing and effect process as the audio buffers could be deleted
            // or modified if an effect is created or deleted
            lockEffectChains_l(effectChains);
        }

        if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
            threadLoop_mix();
        } else {
            threadLoop_sleepTime();
        }

        if (mSuspended > 0) {
            sleepTime = suspendSleepTimeUs();
        }

        // only process effects if we're going to write
        if (sleepTime == 0) {
            for (size_t i = 0; i < effectChains.size(); i ++) {
                effectChains[i]->process_l();
            }
        }

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

        // sleepTime == 0 means we must write to audio hardware
        if (sleepTime == 0) {

            threadLoop_write();

if (mType == MIXER) {
            // write blocked detection
            nsecs_t now = systemTime();
            nsecs_t delta = now - mLastWriteTime;
            if (!mStandby && delta > maxPeriod) {
                mNumDelayedWrites++;
                if ((now - lastWarning) > kWarningThrottleNs) {
#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
                    ScopedTrace st(ATRACE_TAG, "underrun");
#endif
                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
                            ns2ms(delta), mNumDelayedWrites, this);
                    lastWarning = now;
                }
                // FIXME this is broken: longStandbyExit should be handled out of the if() and with
                // a different threshold. Or completely removed for what it is worth anyway...
                if (mStandby) {
                    longStandbyExit = true;
                }
            }
}

            mStandby = false;
        } else {
            usleep(sleepTime);
        }

        // Finally let go of removed track(s), without the lock held
        // since we can't guarantee the destructors won't acquire that
        // same lock.  This will also mutate and push a new fast mixer state.
        threadLoop_removeTracks(tracksToRemove);
        tracksToRemove.clear();

        // FIXME I don't understand the need for this here;
        //       it was in the original code but maybe the
        //       assignment in saveOutputTracks() makes this unnecessary?
        clearOutputTracks();

        // Effect chains will be actually deleted here if they were removed from
        // mEffectChains list during mixing or effects processing
        effectChains.clear();

        // FIXME Note that the above .clear() is no longer necessary since effectChains
        // is now local to this block, but will keep it for now (at least until merge done).
    }

if (mType == MIXER || mType == DIRECT) {
    // put output stream into standby mode
    if (!mStandby) {
        mOutput->stream->common.standby(&mOutput->stream->common);
    }
}
if (mType