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Commit 32584a7d authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Temporary additional logging to investigate bug 

The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.

Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.

Details:
 - include caller pid or client pid where appropriate
 - increase log buffer size
 - log mFastIndex when AudioMixer sees an invalid bufferProvider.
 - log both potentially modified and actually modified tracks in FastMixer.
 - fix benign bug where sq->end() was called more than once.
 - log StateQueue push() call and return.
 - increase StateQueue size from 4 to 8 entries
 - log mixer->enable(), bufferProvider, and currentTrackMask
 - log buffer provider addresses
 - increase fast mixer log buffer again
 - check logf format vs. argument list compatibility
 - add logging to AudioMixer
 - add checking of magic field in AudioMixer to detect overwrites
 - add bool AudioMixer::enabled()
 - increase log buffer sizes yet again
 - enable assertion checking without ALOGV
 - improve a few log messages
 - check for corruption in more places
 - log in all the process hooks
 - add new mixer APIs so we can check for corruption of mixer state
 - fix a build warning

Bug: 6490974
Change-Id: Ib0c4a73dcf606ef9bd898313b3b40ef61ab42f51
parent ab89ac20

include/media/nbaio/NBLog.h

+2 −2
Original line number Diff line number Diff line
@@ -115,7 +115,7 @@ public: 
    virtual ~Writer() { }



    virtual void    log(const char *string);

    virtual void    logf(const char *fmt, ...);

    virtual void    logf(const char *fmt, ...) __attribute__ ((format (printf, 2, 3)));

    virtual void    logvf(const char *fmt, va_list ap);

    virtual void    logTimestamp();

    virtual void    logTimestamp(const struct timespec& ts);
@@ -149,7 +149,7 @@ public: 
    LockedWriter(size_t size, void *shared);



    virtual void    log(const char *string);

    virtual void    logf(const char *fmt, ...);

    virtual void    logf(const char *fmt, ...) __attribute__ ((format (printf, 2, 3)));

    virtual void    logvf(const char *fmt, va_list ap);

    virtual void    logTimestamp();

    virtual void    logTimestamp(const struct timespec& ts);

services/audioflinger/AudioFlinger.h

+1 −1
Original line number Diff line number Diff line
@@ -227,7 +227,7 @@ public: 
    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);

    void                unregisterWriter(const sp<NBLog::Writer>& writer);

private:

    static const size_t kLogMemorySize = 10 * 1024;

    static const size_t kLogMemorySize = 50 * 1024;

    sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled

public:

services/audioflinger/AudioMixer.cpp

+68 −4
Original line number Diff line number Diff line
@@ -16,7 +16,7 @@ 
*/



#define LOG_TAG "AudioMixer"

//#define LOG_NDEBUG 0

#define LOG_NDEBUG 0



#include <stdint.h>

#include <string.h>
@@ -25,6 +25,8 @@ 


#include <utils/Errors.h>

#include <utils/Log.h>

#undef ALOGV

#define ALOGV(a...) do { } while (0)



#include <cutils/bitops.h>

#include <cutils/compiler.h>
@@ -98,7 +100,7 @@ effect_descriptor_t AudioMixer::dwnmFxDesc; 


AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)

    :   mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),

        mSampleRate(sampleRate)

        mSampleRate(sampleRate), mLog(&mDummyLog)

{

    // AudioMixer is not yet capable of multi-channel beyond stereo

    COMPILE_TIME_ASSERT_FUNCTION_SCOPE(2 == MAX_NUM_CHANNELS);
@@ -122,6 +124,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr 
    mState.hook         = process__nop;

    mState.outputTemp   = NULL;

    mState.resampleTemp = NULL;

    mState.mLog         = &mDummyLog;

    // mState.reserved



    // FIXME Most of the following initialization is probably redundant since
@@ -131,6 +134,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr 
    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {

        t->resampler = NULL;

        t->downmixerBufferProvider = NULL;

        t->magic = track_t::kMagic;

        t++;

    }
@@ -169,6 +173,12 @@ AudioMixer::~AudioMixer() 
    delete [] mState.resampleTemp;

}



void AudioMixer::setLog(NBLog::Writer *log)

{

    mLog = log;

    mState.mLog = log;

}



int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)

{

    uint32_t names = (~mTrackNames) & mConfiguredNames;
@@ -209,9 +219,12 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) 
        t->mainBuffer = NULL;

        t->auxBuffer = NULL;

        t->downmixerBufferProvider = NULL;

        t->fastIndex = -1;

        // t->magic unchanged



        status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);

        if (status == OK) {

            mLog->logf("getTrackName %d", n);

            return TRACK0 + n;

        }

        ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix",
@@ -366,9 +379,11 @@ void AudioMixer::deleteTrackName(int name) 
{

    ALOGV("AudioMixer::deleteTrackName(%d)", name);

    name -= TRACK0;

    mLog->logf("deleteTrackName %d", name);

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    ALOGV("deleteTrackName(%d)", name);

    track_t& track(mState.tracks[ name ]);

    track.checkMagic();

    if (track.enabled) {

        track.enabled = false;

        invalidateState(1<<name);
@@ -387,8 +402,10 @@ void AudioMixer::enable(int name) 
    name -= TRACK0;

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    track_t& track = mState.tracks[name];

    track.checkMagic();



    if (!track.enabled) {

        mLog->logf("enable %d", name);

        track.enabled = true;

        ALOGV("enable(%d)", name);

        invalidateState(1 << name);
@@ -400,19 +417,36 @@ void AudioMixer::disable(int name) 
    name -= TRACK0;

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    track_t& track = mState.tracks[name];

    track.checkMagic();



    if (track.enabled) {

        mLog->logf("disable %d", name);

        track.enabled = false;

        ALOGV("disable(%d)", name);

        invalidateState(1 << name);

    }

}



bool AudioMixer::enabled(int name)

{

    name -= TRACK0;

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    track_t& track = mState.tracks[name];

    track.checkMagic();

#if 0

    // can't do this because mState.enabledTracks is updated lazily

    ALOG_ASSERT(track.enabled == ((mState.enabledTracks & (1 << name)) != 0));

#endif



    return track.enabled;

}



void AudioMixer::setParameter(int name, int target, int param, void *value)

{

    name -= TRACK0;

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    track_t& track = mState.tracks[name];

    track.checkMagic();



    int valueInt = (int)value;

    int32_t *valueBuf = (int32_t *)value;
@@ -455,6 +489,9 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) 
        //         for a specific track? or per mixer?

        /* case DOWNMIX_TYPE:

            break          */

        case FAST_INDEX:

            track.fastIndex = valueInt;

            break;

        default:

            LOG_FATAL("bad param");

        }
@@ -540,6 +577,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) 


bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)

{

    checkMagic();

    if (value != devSampleRate || resampler != NULL) {

        if (sampleRate != value) {

            sampleRate = value;
@@ -572,6 +610,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) 
inline

void AudioMixer::track_t::adjustVolumeRamp(bool aux)

{

    checkMagic();

    for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) {

        if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||

            ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
@@ -600,8 +639,10 @@ size_t AudioMixer::getUnreleasedFrames(int name) const 
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)

{

    name -= TRACK0;

    mLog->logf("set bp %d=%p", name, bufferProvider);

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);



    mState.tracks[name].checkMagic();

    if (mState.tracks[name].downmixerBufferProvider != NULL) {

        // update required?

        if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) {
@@ -619,10 +660,27 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider 
    }

}



AudioBufferProvider* AudioMixer::getBufferProvider(int name)

{

    name -= TRACK0;

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    mState.tracks[name].checkMagic();

    return mState.tracks[name].bufferProvider;

}



int AudioMixer::getFastIndex(int name)

{

    name -= TRACK0;

    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);

    mState.tracks[name].checkMagic();

    return mState.tracks[name].fastIndex;

}



void AudioMixer::process(int64_t pts)

{

    if (mState.needsChanged) {

        mLog->logf("process needs=%#x", mState.needsChanged);

    }

    mState.hook(&mState, pts);

}
@@ -647,6 +705,7 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) 
    }

    state->enabledTracks &= ~disabled;

    state->enabledTracks |=  enabled;

    state->mLog->logf("process_validate ena=%#x", state->enabledTracks);



    // compute everything we need...

    int countActiveTracks = 0;
@@ -1058,6 +1117,7 @@ void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, 
void AudioMixer::process__nop(state_t* state, int64_t pts)

{

    uint32_t e0 = state->enabledTracks;

    state->mLog->logf("process_nop ena=%#x", e0);

    size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;

    while (e0) {

        // process by group of tracks with same output buffer to
@@ -1103,6 +1163,7 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 


    // acquire each track's buffer

    uint32_t enabledTracks = state->enabledTracks;

    state->mLog->logf("process_gNR ena=%#x", enabledTracks);

    uint32_t e0 = enabledTracks;

    while (e0) {

        const int i = 31 - __builtin_clz(e0);
@@ -1111,8 +1172,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 
        t.buffer.frameCount = state->frameCount;

        int valid = t.bufferProvider->getValid();

        if (valid != AudioBufferProvider::kValid) {

            ALOGE("invalid bufferProvider=%p name=%d frameCount=%d valid=%#x enabledTracks=%#x",

                    t.bufferProvider, i, t.buffer.frameCount, valid, enabledTracks);

            ALOGE("invalid bufferProvider=%p name=%d fastIndex=%d frameCount=%d valid=%#x enabledTracks=%#x",

                    t.bufferProvider, i, t.fastIndex, t.buffer.frameCount, valid, enabledTracks);

            // expect to crash

        }

        t.bufferProvider->getNextBuffer(&t.buffer, pts);
@@ -1211,6 +1272,7 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 
    size_t numFrames = state->frameCount;



    uint32_t e0 = state->enabledTracks;

    state->mLog->logf("process_gR ena=%#x", e0);

    while (e0) {

        // process by group of tracks with same output buffer

        // to optimize cache use
@@ -1275,6 +1337,7 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,

                                                           int64_t pts)

{

    state->mLog->logf("process_1TSNR ena=%#x", state->enabledTracks);

    // This method is only called when state->enabledTracks has exactly

    // one bit set.  The asserts below would verify this, but are commented out

    // since the whole point of this method is to optimize performance.
@@ -1344,6 +1407,7 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, 
{

    int i;

    uint32_t en = state->enabledTracks;

    state->mLog->logf("process_2TSNR ena=%#x", en);



    i = 31 - __builtin_clz(en);

    const track_t& t0 = state->tracks[i];

services/audioflinger/AudioMixer.h

+23 −2
Original line number Diff line number Diff line
@@ -28,6 +28,7 @@ 


#include <audio_effects/effect_downmix.h>

#include <system/audio.h>

#include <media/nbaio/NBLog.h>



namespace android {
@@ -76,6 +77,7 @@ public: 
        MAIN_BUFFER     = 0x4002,

        AUX_BUFFER      = 0x4003,

        DOWNMIX_TYPE    = 0X4004,

        FAST_INDEX      = 0x4005, // for debugging only

        // for target RESAMPLE

        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;

                                  // parameter 'value' is the new sample rate in Hz.
@@ -106,13 +108,17 @@ public: 
    // Enable or disable an allocated track by name

    void        enable(int name);

    void        disable(int name);

    bool        enabled(int name);



    void        setParameter(int name, int target, int param, void *value);



    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);

    AudioBufferProvider* getBufferProvider(int name);

    void        process(int64_t pts);



    uint32_t    trackNames() const { return mTrackNames; }

    uint32_t    enabledTrackNames() const { return mState.enabledTracks; }

    int         getFastIndex(int name);



    size_t      getUnreleasedFrames(int name) const;
@@ -200,7 +206,10 @@ private: 


        int32_t     sessionId;



        int32_t     padding[2];

        int32_t     fastIndex;

        int32_t     magic;

        static const int kMagic = 0x54637281;

        //int32_t     padding[1];



        // 16-byte boundary
@@ -210,6 +219,12 @@ private: 
        void        adjustVolumeRamp(bool aux);

        size_t      getUnreleasedFrames() const { return resampler != NULL ?

                                                    resampler->getUnreleasedFrames() : 0; };

        void        checkMagic() {

            if (magic != kMagic) {

                ALOGE("magic=%#x fastIndex=%d", magic, fastIndex);

            }

        }



    };



    // pad to 32-bytes to fill cache line
@@ -220,7 +235,8 @@ private: 
        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL

        int32_t         *outputTemp;

        int32_t         *resampleTemp;

        int32_t         reserved[2];

        NBLog::Writer*  mLog;

        int32_t         reserved[1];

        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS

        track_t         tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));

    };
@@ -247,6 +263,11 @@ private: 


    const uint32_t  mSampleRate;



    NBLog::Writer*  mLog;

    NBLog::Writer   mDummyLog;

public:

    void            setLog(NBLog::Writer* log);

private:

    state_t         mState __attribute__((aligned(32)));



    // effect descriptor for the downmixer used by the mixer

services/audioflinger/FastMixer.cpp

+89 −4
Original line number Diff line number Diff line
@@ -21,13 +21,15 @@ 
// </IMPORTANT_WARNING>



#define LOG_TAG "FastMixer"

//#define LOG_NDEBUG 0

#define LOG_NDEBUG 0



#define ATRACE_TAG ATRACE_TAG_AUDIO



#include <sys/atomics.h>

#include <time.h>

#include <utils/Log.h>

#undef ALOGV

#define ALOGV(a...) do { } while (0)

#include <utils/Trace.h>

#include <system/audio.h>

#ifdef FAST_MIXER_STATISTICS
@@ -93,6 +95,8 @@ bool FastMixer::threadLoop() 
    uint32_t warmupCycles = 0;  // counter of number of loop cycles required to warmup

    NBAIO_Sink* teeSink = NULL; // if non-NULL, then duplicate write() to this non-blocking sink

    NBLog::Writer dummyLogWriter, *logWriter = &dummyLogWriter;

    bool myEnabled[FastMixerState::kMaxFastTracks];

    memset(myEnabled, 0, sizeof(myEnabled));



    for (;;) {
@@ -120,12 +124,16 @@ bool FastMixer::threadLoop() 
        FastMixerState::Command command = next->mCommand;

        if (next != current) {



            logWriter->logTimestamp();

            logWriter->log("next != current");



            // As soon as possible of learning of a new dump area, start using it

            dumpState = next->mDumpState != NULL ? next->mDumpState : &dummyDumpState;

            teeSink = next->mTeeSink;

            logWriter = next->mNBLogWriter != NULL ? next->mNBLogWriter : &dummyLogWriter;

            if (mixer != NULL) {

                mixer->setLog(logWriter);

            }



            // We want to always have a valid reference to the previous (non-idle) state.

            // However, the state queue only guarantees access to current and previous states.
@@ -300,13 +308,21 @@ bool FastMixer::threadLoop() 
                    addedTracks &= ~(1 << i);

                    const FastTrack* fastTrack = &current->mFastTracks[i];

                    AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;

                    logWriter->logf("bp %d i=%d %p", __LINE__, i, bufferProvider);

                    ALOG_ASSERT(bufferProvider != NULL && fastTrackNames[i] == -1);

                    if (bufferProvider == NULL ||

                            bufferProvider->getValid() != AudioBufferProvider::kValid) {

                        logWriter->logTimestamp();

                        logWriter->logf("added invalid %#x", i);

                    }

                    if (mixer != NULL) {

                        // calling getTrackName with default channel mask and a random invalid

                        //   sessionId (no effects here)

                        name = mixer->getTrackName(AUDIO_CHANNEL_OUT_STEREO, -555);

                        ALOG_ASSERT(name >= 0);

                        fastTrackNames[i] = name;

                        mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FAST_INDEX,

                                (void *) i);

                        mixer->setBufferProvider(name, bufferProvider);

                        mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,

                                (void *) mixBuffer);
@@ -317,27 +333,41 @@ bool FastMixer::threadLoop() 
                        }

                        mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,

                                (void *) fastTrack->mChannelMask);

                        if (!mixer->enabled(name)) {

                            logWriter->logf("enable %d i=%d name=%d", __LINE__, i, name);

                        }

                        mixer->enable(name);

                        myEnabled[i] = true;

                    }

                    generations[i] = fastTrack->mGeneration;

                }



                // finally process modified tracks; these use the same slot

                // finally process (potentially) modified tracks; these use the same slot

                // but may have a different buffer provider or volume provider

                unsigned modifiedTracks = currentTrackMask & previousTrackMask;

                if (modifiedTracks) {

                    logWriter->logf("modified %#x", modifiedTracks);

                    logWriter->logf("pot. mod. %#x", modifiedTracks);

                }

                unsigned actuallyModifiedTracks = 0;

                while (modifiedTracks != 0) {

                    i = __builtin_ctz(modifiedTracks);

                    modifiedTracks &= ~(1 << i);

                    const FastTrack* fastTrack = &current->mFastTracks[i];

                    if (fastTrack->mGeneration != generations[i]) {

                        actuallyModifiedTracks |= 1 << i;

                        AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;

                        logWriter->logf("bp %d i=%d %p", __LINE__, i, bufferProvider);

                        ALOG_ASSERT(bufferProvider != NULL);

                        if (bufferProvider == NULL ||

                                bufferProvider->getValid() != AudioBufferProvider::kValid) {

                            logWriter->logTimestamp();

                            logWriter->logf("modified invalid %#x", i);

                        }

                        if (mixer != NULL) {

                            name = fastTrackNames[i];

                            ALOG_ASSERT(name >= 0);

                            mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FAST_INDEX,

                                    (void *) i);

                            mixer->setBufferProvider(name, bufferProvider);

                            if (fastTrack->mVolumeProvider == NULL) {

                                mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
@@ -360,6 +390,9 @@ bool FastMixer::threadLoop() 
                        generations[i] = fastTrack->mGeneration;

                    }

                }

                if (actuallyModifiedTracks) {

                    logWriter->logf("act. mod. %#x", actuallyModifiedTracks);

                }



                fastTracksGen = current->mFastTracksGen;
@@ -377,6 +410,7 @@ bool FastMixer::threadLoop() 
            ALOG_ASSERT(mixBuffer != NULL);

            // for each track, update volume and check for underrun

            unsigned currentTrackMask = current->mTrackMask;

            logWriter->logf("ctm %#x", currentTrackMask);

            while (currentTrackMask != 0) {

                i = __builtin_ctz(currentTrackMask);

                currentTrackMask &= ~(1 << i);
@@ -410,25 +444,76 @@ bool FastMixer::threadLoop() 
                        underruns.mBitFields.mEmpty++;

                        underruns.mBitFields.mMostRecent = UNDERRUN_EMPTY;

                        mixer->disable(name);

                        myEnabled[i] = false;

                    } else {

                        // allow mixing partial buffer

                        underruns.mBitFields.mPartial++;

                        underruns.mBitFields.mMostRecent = UNDERRUN_PARTIAL;

                        if (!mixer->enabled(name)) {

                            logWriter->logf("enable %d i=%d name=%d", __LINE__, i, name);

                        }

                        mixer->enable(name);

                        myEnabled[i] = true;

                    }

                } else {

                    underruns.mBitFields.mFull++;

                    underruns.mBitFields.mMostRecent = UNDERRUN_FULL;

                    if (!mixer->enabled(name)) {

                        logWriter->logf("enable %d i=%d name=%d", __LINE__, i, name);

                    }

                    mixer->enable(name);

                    myEnabled[i] = true;

                }

                ftDump->mUnderruns = underruns;

                ftDump->mFramesReady = framesReady;

                AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;

                if (bufferProvider == NULL ||

                        bufferProvider->getValid() != AudioBufferProvider::kValid) {

                    logWriter->logTimestamp();

                    logWriter->logf("mixing invalid %#x", i);

                }

            }



            int64_t pts;

            if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts)))

                pts = AudioBufferProvider::kInvalidPTS;



            // validate that mixer state is correct

            currentTrackMask = current->mTrackMask;

            unsigned expectedMixerEnabled = 0;

            while (currentTrackMask != 0) {

                i = __builtin_ctz(currentTrackMask);

                currentTrackMask &= ~(1 << i);

                const FastTrack* fastTrack = &current->mFastTracks[i];

                int name = fastTrackNames[i];

                ALOG_ASSERT(name >= 0);

                bool isEnabled = mixer->enabled(name);

                if (isEnabled != myEnabled[i]) {

                    logWriter->logTimestamp();

                    logWriter->logf("? i=%d name=%d mixena=%d ftena=%d", i, name, isEnabled,

                            myEnabled[i]);

                }

                if (myEnabled[i]) {

                    expectedMixerEnabled |= 1 << name;

                }

                AudioBufferProvider *abp = mixer->getBufferProvider(name);

                if (abp != fastTrack->mBufferProvider) {

                    logWriter->logTimestamp();

                    logWriter->logf("? i=%d name=%d mixbp=%p ftbp=%p", i, name, abp,

                            fastTrack->mBufferProvider);

                }

                int fastIndex = mixer->getFastIndex(name);

                if (fastIndex != (int) i) {

                    logWriter->logTimestamp();

                    logWriter->logf("? i=%d name=%d fastIndex=%d", i, name, fastIndex);

                }

            }

            unsigned mixerEnabled = mixer->enabledTrackNames();

            if (mixerEnabled != expectedMixerEnabled) {

                logWriter->logTimestamp();

                logWriter->logf("? mixena=%#x expected=%#x", mixerEnabled, expectedMixerEnabled);

            }



            // process() is CPU-bound

            mixer->process(pts);

            mixBufferState = MIXED;
@@ -453,7 +538,7 @@ bool FastMixer::threadLoop() 
            ATRACE_END();

            dumpState->mWriteSequence++;

            if (framesWritten >= 0) {

                ALOG_ASSERT(framesWritten <= frameCount);

                ALOG_ASSERT((size_t) framesWritten <= frameCount);

                dumpState->mFramesWritten += framesWritten;

                //if ((size_t) framesWritten == frameCount) {

                //    didFullWrite = true;