SoundFile.cpp

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//
//  Abst_SoundFile.cpp -- CSL's abstract sound file class
//  See the copyright notice and acknowledgment of authors in the file COPYRIGHT
//

#include "SoundFile.h"

#ifdef USE_TAGS
#include <taglib/fileref.h>         // libTag includes
#include <taglib/tag.h>
#endif

using namespace csl;

SoundFileMetadata::SoundFileMetadata() { }

void SoundFileMetadata::dump() {
    if (mTitle.empty()) {
        logMsg("No metadata");
        return;
    }
    int secs = mLength % 60;                        // make a mins:secs display for the duration
    int mins = (mLength - secs) / 60;
    char dur[CSL_WORD_LEN];
    if (secs < 10)
        sprintf(dur, "%d:0%d", mins, secs);
    else
        sprintf(dur, "%d:%d", mins, secs);
    logMsg("Tags: \"%s\" - %s - %s - %d - %s min", 
            mTitle.c_str(), mArtist.c_str(), mAlbum.c_str(), mYear, dur);
    logMsg("        trk %d - %s - %d kbps - %d Hz - %d ch", 
            mTrack, mGenre.c_str(), mBitRate, mSampleRate, mChannels);
//  if ( ! mComment.empty()) 
//      logMsg("        Comment: %s", mComment.c_str());
}

// Abst_SoundFile Constructors

Abst_SoundFile::Abst_SoundFile(string tpath, int tstart, int tstop) : 
            WavetableOscillator(), Writeable(), Seekable(), 
            mProperties(0),
            mPath(string(tpath)), 
            mMode(kSoundFileClosed),
            mFormat(kSoundFileFormatOther),
            mIsValid(false), 
            mIsLooping(false),
            mStart(tstart), 
            mStop(tstop),
            mRate(1.0),
            mNumFrames(0),
            mBytesPerSample(0),
            mBase(0) { /* no-op */}

///< Copy constructor -- shares sample buffer

Abst_SoundFile::Abst_SoundFile(Abst_SoundFile & otherSndFile) :
            WavetableOscillator(), Writeable(), Seekable(), 
            mProperties(otherSndFile.mProperties),
            mMode(otherSndFile.mode()), 
            mIsValid(otherSndFile.isValid()), 
            mIsLooping(otherSndFile.isLooping()),
            mStart(otherSndFile.startFrame()), 
            mStop(otherSndFile.stopFrame()),
            mRate(otherSndFile.playbackRate()),
            mBase(otherSndFile.mBase) {
    this->setWaveform(otherSndFile.mWavetable), mCurrentFrame = 0;
    if ( ! otherSndFile.isCached())
        logMsg(kLogError, "Cannot copy uncached sound file \"%s\"", mPath.c_str());
    setPath(otherSndFile.path());
}

// Clean up data allocated

Abst_SoundFile::~Abst_SoundFile() {
//  freeBuffer();       this is done by the WaveTableOsc destructor
    if (mProperties) 
        delete mProperties;
}

void Abst_SoundFile::setPath(string tpath) {
    mPath = tpath;
}

unsigned Abst_SoundFile::channels() const {
    return mIsValid ? mNumChannels : 0;
}

float Abst_SoundFile::durationInSecs() {
    return mIsValid ? ((float) (mStop - mStart) / (float) mFrameRate) : 0.0;
}

void Abst_SoundFile::freeBuffer() {
    if (mWavetable.mAreBuffersAllocated)            // if no buffer allocated
        mWavetable.freeBuffers();
}

// check if the buffer's big enough

void Abst_SoundFile::checkBuffer(unsigned numFrames) {
    if ( ! mWavetable.mAreBuffersAllocated) {               // if no sample read buffer allocated
        mWavetable.freeBuffers();
        mWavetable.setSize(mNumChannels, numFrames);
        mWavetable.allocateBuffers();
    } else if (mWavetable.mNumFrames < numFrames) {         // if asking for more samples than fit in the buffer
        logMsg("Reallocating sound file buffers (%d)", numFrames * mNumChannels);   
        mWavetable.freeBuffers();
        mWavetable.setSize(mNumChannels, numFrames);
        mWavetable.allocateBuffers();
    }
//#ifdef CSL_USE_SRConv
//  if ( ! mSRConvBuffer.mAreBuffersAllocated) {                    // if no SRC buffer allocated
//      mSRConvBuffer.setSize(1, CGestalt::maxBufferFrames() * mNumChannels);
//      mSRConvBuffer.allocateBuffers();
//  }
//#endif
}

void Abst_SoundFile::checkBuffer(unsigned numChans, unsigned numFrames) {
    if ( ! mWavetable.mAreBuffersAllocated) {               // if no sample read buffer allocated
        mNumChannels = numChans;
        mWavetable.setSize(mNumChannels, numFrames);
        mWavetable.allocateBuffers();
                            // if asking for more samples than fit in the buffer
    } else if ((mWavetable.mNumFrames < numFrames) || (mNumChannels != numChans)) {
        logMsg("Reallocating sound file buffers (%d)", numFrames * mNumChannels);   
        mNumChannels = numChans;
        mWavetable.freeBuffers();
        mWavetable.setSize(mNumChannels, numFrames);
        mWavetable.allocateBuffers();
    }
}

// average all the channels to mono

void Abst_SoundFile::mergeToMono() {
    if ( ! mWavetable.mAreBuffersAllocated) {
        logMsg("Abst_SoundFile::mergeToMono - buffers not allocated");  
        return;
    }
    if (mNumChannels == 1) {
        logMsg("Abst_SoundFile::mergeToMono - sound file already mono");    
        return;
    }
    Buffer newBuf(1, mWavetable.mNumFrames);    // create a new mono buffer
    newBuf.allocateBuffers();
                                                // sum channels
    unsigned numCh = mNumChannels;
    for (unsigned i = 0; i < mWavetable.mNumFrames; i++) {
        sample sum = 0.0f;
        for (unsigned j = 0; j < numCh; j++)
            sum += (mWavetable.buffer(j))[i];
                                                // store the average
        newBuf.setBuffer(0, i, sum / numCh);
    }                                           // now overwrite my own mWavetable
    mWavetable.copyFrom(newBuf);
    newBuf.mDidIAllocateBuffers = false;
    mWavetable.mDidIAllocateBuffers = true;
    mNumChannels = 1;
}

// Abst_SoundFile::convertRate - perform sample-rate conversion

void Abst_SoundFile::convertRate(int fromRate, int toRate) {
#ifndef USE_SRC
    return;
#else
    mWavetable.convertRate(fromRate, toRate);
    mFrameRate = toRate;
    mNumFrames = mWavetable.mNumFrames;
#endif
}

// ~~~~~~~ Accessors ~~~~~~~~

void Abst_SoundFile::setStart(int val) { 
    mStart = val;
    if (mStart < 0) 
        mStart = 0;
    if ((unsigned) mStart >= duration()) 
        mStart = (int) duration();
    if (mIsValid)
        seekTo(mStart);
}

void Abst_SoundFile::setStartSec(float val) {
    setStart((int) (val * mFrameRate));
}

void Abst_SoundFile::setStartRatio(float val) {
    setStart((int) (val * duration()));
}

void Abst_SoundFile::setStop(int val) { 
    mStop = val;
    if (mStop < 0) 
        mStop = 0;
    int du = (int) duration();
    if (mStop > du) 
        mStop = du;
}

void Abst_SoundFile::setStopSec(float val) {
    setStop((int) (val * mFrameRate));
}

void Abst_SoundFile::setStopRatio(float val) {
    setStop((int) (val * duration()));
}

void Abst_SoundFile::setBase(int val) {
    mBase = val;
}

// Rate and transposition

void Abst_SoundFile::setRate(UnitGenerator & frequency) { 
    if ( ! mInputs[CSL_RATE])
        this->addInput(CSL_RATE, frequency);
    else
        mInputs[CSL_RATE]->mUGen = & frequency;
}

void Abst_SoundFile::setRate(float frequency) { 
    mRate = frequency;
#ifdef CSL_DEBUG
    logMsg("FrequencyAmount::set scale input value");
#endif
}

bool Abst_SoundFile::isActive() {
    if ( ! mIsValid)
        return false;
    return (mCurrentFrame < (unsigned) mStop);
}

bool Abst_SoundFile::isCached() { 
    return (mWavetable.mNumFrames == duration()); 
}

///< answer if file has X samples in RAM

bool Abst_SoundFile::isCached(unsigned samps) { 
    return (mWavetable.mNumFrames < (mCurrentFrame + samps));
}

// trigger the file to start

void Abst_SoundFile::trigger() {
    if (mStart > 0) {
        mCurrentFrame = mStart;
        seekTo(mStart);
    } else {
        mCurrentFrame = 0;
        seekTo(0);
    }
}

// set the pointer to the end of the file

void Abst_SoundFile::setToEnd() {
    mCurrentFrame = mStop;
}

// Read the ID3 tags. Returns true if able to read them from the file

bool Abst_SoundFile::readTags() throw (CException) {
#ifdef USE_TAGS
    if ( ! mProperties) 
        mProperties = new SoundFileMetadata;
    TagLib::FileRef tfil(mPath.c_str());
    if ( ! tfil.isNull() && tfil.tag()) {
        TagLib::Tag *tag = tfil.tag();
        mProperties->mTitle     = string(tag->title().toCString());
        mProperties->mArtist    = string(tag->artist().toCString());
        mProperties->mAlbum     = string(tag->album().toCString());
        mProperties->mYear      = tag->year();
        mProperties->mComment   = string(tag->comment().toCString());
        mProperties->mTrack     = tag->track();
        mProperties->mGenre     = string(tag->genre().toCString());
    }
    if( ! tfil.isNull() && tfil.audioProperties()) {
        TagLib::AudioProperties *properties = tfil.audioProperties();
        mProperties->mBitRate   = properties->bitrate();
        mProperties->mSampleRate= properties->sampleRate();
        mProperties->mChannels  = properties->channels();
        mProperties->mLength    = properties->length();
    }
    return !tfil.isNull();
#else
    return false;
#endif
}

// log snd file props

void Abst_SoundFile::dump() {
    const char * nam = path().c_str();
    if (strlen(nam) > 50)
        logMsg("SndFile \"%s\"\n\t\t%d Hz, %d ch, %5.3f sec", 
                nam, frameRate(), channels(), durationInSecs());
    else
        logMsg("SndFile \"%s\" - %d Hz, %d ch, %5.3f sec", 
                nam, frameRate(), channels(), durationInSecs());
}

////////////// next_buffer -- the work is done here //////////

void Abst_SoundFile::nextBuffer(Buffer &outputBuffer) throw(CException) {
    unsigned numFrames = outputBuffer.mNumFrames;
    unsigned currentFrame = mCurrentFrame;
    DECLARE_SCALABLE_CONTROLS;                      // declare the scale/offset buffers and values

    if (currentFrame >= (unsigned) mStop) {         // if done
        outputBuffer.zeroBuffers();
        return;
    }
    if (currentFrame + numFrames >= (unsigned) mStop) {     // if final window
        numFrames = mStop - currentFrame;
        outputBuffer.zeroBuffers();
    }
    if ( ! this->isCached(numFrames)) {             // if not playing from cache buffer
        this->readBufferFromFile(numFrames);        // read from file
    }

    LOAD_SCALABLE_CONTROLS;                         // load the scaleC and offsetC from the constant or dynamic value

    if (mRate == 1) {                                               // if playing at normal rate
        if (scalePort->isFixed() && offsetPort->isFixed()           // if fixed scale/offset, use memcpy
                    && (scaleValue == 1) && (offsetValue == 0)) {
            unsigned numBytes = numFrames * sizeof(sample);         // buffer copy loop
            for (unsigned i = 0; i < outputBuffer.mNumChannels; i++) {
                int which = csl_min(i, (mNumChannels - 1));
                SampleBuffer sndPtr = mWavetable.monoBuffer(which) + currentFrame;
                SampleBuffer outPtr = outputBuffer.monoBuffer(i);
                memcpy(outPtr, sndPtr, numBytes);                   // here's the memcpy
            }
        } else {                                                    // else loop applying scale/offset
            sample samp;
            for (unsigned i = 0; i < outputBuffer.mNumChannels; i++) {
                SampleBuffer buffer = outputBuffer.monoBuffer(i);   // get pointer to the selected output channel
                SampleBuffer dPtr = mWavetable.monoBuffer(csl_min(i, (mNumChannels - 1))) + currentFrame;
                for (unsigned j = 0; j < numFrames; j++) {          // here's the sample loop
                    samp = (*dPtr++ * scaleValue) + offsetValue;    // get and scale the file sample
                    *buffer++ = samp;
                    UPDATE_SCALABLE_CONTROLS;                       // update the dynamic scale/offset
                }
                scalePort->resetPtr();
                offsetPort->resetPtr();
            }
        }
    } else {                                        // if mRate != 1.0, 
                                                    // use wavetable interpolation
        this->setFrequency(mRate);
        for (unsigned i = 0; i < mNumChannels; i++)
            WavetableOscillator::nextBuffer(outputBuffer, i);   
    }
    currentFrame += numFrames;                              // increment buf ptr
    if ((currentFrame >= (unsigned) mStop) && mIsLooping)   // if we are past the end of the file...
        currentFrame = 0;                           // this will click, have to call nextBuffer() recursively here
    mCurrentFrame = currentFrame;                   // store back to member
    return;
}


////////////////////////////////////////////////////////////////////////////////////////////
//
// Sound Cue implementation
//

// Generic constructors

SoundCue::SoundCue() {
    mFile = NULL;
    mCurrent = 0;
}

SoundCue::SoundCue(string name, Abst_SoundFile *file, int start, int stop) : 
            mName(name), 
            mFile(file), 
            mStart(start), 
            mStop(stop) { 
    mCurrent = mStart;
    mReadRate = (UnitGenerator*) 0;
}

SoundCue::~SoundCue() { }

// Read an instance's data from a file
// The format is that used in the SeSpSp layer index file:
//   name      start    stop
//  b5b.r21e.aiff 13230080 15876096     -- NB: these are in FRAMES

void SoundCue::readFrom (FILE *input) {
    char cmName[128];
    unsigned start, stop;

    if(fscanf(input, "%s %u %u\n", cmName, &start, &stop) == 3) {
        mName = cmName;
        mStart = start;
        mStop = stop;
        mCurrent = mStart;
    }
}

// Pretty-print the receiver

void SoundCue::dump() {
    logMsg("\tSC: \"%s\" %d - %d (%.3f)\n", 
            mName.c_str(), mStart, mStop, 
            ((float)(mStop - mStart) / (float)(mFile->frameRate() * mFile->channels())));
}

// Copy samples from the file's sample buffer to the output
// I assume that the # of channels in the output is the same as in my file, i.e.,
// To play a mono file, "wrap" it in a Deinterleaver

void SoundCue::nextBuffer(Buffer &outputBuffer) throw(CException) {
    unsigned numFrames = outputBuffer.mNumFrames;
    unsigned toCopy = 0, toZero;
    SampleBuffer out;
    
    for (unsigned h = 0; h < outputBuffer.mNumChannels; h++) {
        out = outputBuffer.buffer(h);
        if (mCurrent >= mStop) {            // if done
            for (unsigned i = 0; i < numFrames; i++) 
                *out++ = 0.0;
            continue;
        }   
        SampleBuffer samps = (mFile->mWavetable.monoBuffer(0)) + (mCurrent * mFile->channels()) + h;
        if ( ! mFile->isValid()) {      // if no file data
            fprintf(stderr, "\tCannot play uncached sound file \"%s\"n", mName.c_str());
            for (unsigned i = 0; i < numFrames; i++) 
                *out++ = 0.0;
            continue;
        }
        toCopy = numFrames;     // figure out how many samples to copy
        toZero = 0; 
        if ((mCurrent + numFrames) >= (unsigned) mStop) {
            toCopy = mStop - mCurrent;
            toZero = numFrames - toCopy;
        //  fprintf(stderr, "\tSample \"%s\" finished\n", mName);
        }                       // Now do the copy loops
        for (unsigned i = 0; i < toCopy; i++) 
            *out++ = *samps++;
        for (unsigned i = 0; i < toZero; i++) 
            *out++ = 0.0;
    }
    mCurrent += toCopy;         // increment the receiver's sample pointer
    return;
}

// Utility functions

bool SoundCue::isActive() { 
    if (mFile == 0)
        return (false);
    return (mCurrent < mStop);
}

void SoundCue::setToEnd(void) {
    mCurrent = mStop;
}

void SoundCue::trigger(void) {
    mCurrent = mStart;
    mFloatCurrent = 0.0;
}

#ifdef USE_SNDFILEBUFFER

////////////////////////////////////////////////////////////////////////////////////////////
//
// SoundFileBuffer implementation
//

#include "SoundFileL.h"         // abstract class header


SoundFileBuffer::SoundFileBuffer(string path, unsigned numFrames) 
        : Buffer(CGestalt::numOutChannels(), numFrames) {
    mFile = (Abst_SoundFile * ) (new LSoundFile(path));
    mFile->openForRead(false);
    mFile->readBufferFromFile(CGestalt::sndFileFrames());
    mNumChannels = mFile->channels();
    mNumFrames = mFile->duration();
    mNumAlloc = csl_min(CGestalt::sndFileFrames(), mNumFrames);
    mMonoBufferByteSize = mNumFrames * mNumChannels * sizeof(sample);
    mAreBuffersAllocated = true;
    mDidIAllocateBuffers = false;
    
    mBuffers = new SampleBuffer[mNumChannels];  // reserve space for buffers
    for (unsigned i = 0; i < mNumChannels; i++)
        setBuffer(i, mFile->buffer(i));
}

///< Copy constructor -- shares sample buffer

SoundFileBuffer::SoundFileBuffer(Abst_SoundFile & otherSndFile) {
    // setBuffer
}

SoundFileBuffer::~SoundFileBuffer() { }


/// answer a samp ptr, testing that it's in RAM

sample * SoundFileBuffer::samplePtrFor(unsigned channel, unsigned offset){
    if ((mFile->base() <= offset) && (mFile->base() + mFile->cacheSize() > offset))
        return(mFile->mWavetable.buffer(channel) + (offset - mFile->base()));
    if (mFile) {
        mFile->seekTo(offset);
        mFile->readBufferFromFile(mFile->mWavetable.mNumFrames);
        return(mFile->mWavetable.buffer(channel));
    }
    return NULL;
}

/// answer a samp ptr tested for extent (offset + maxFrame)

sample * SoundFileBuffer::samplePtrFor(unsigned channel, unsigned offset, unsigned maxFrame){
    if ((mFile->base() <= offset) && ((mFile->base() + mNumAlloc) > (offset + maxFrame)))
        return (mFile->mWavetable.buffer(channel) + (offset - mFile->base()));
    if (mFile) {
        mFile->seekTo(offset);
        mFile->readBufferFromFile(mFile->mWavetable.mNumFrames);
        return(mFile->mWavetable.buffer(channel));
    }
    return NULL;
}

#endif

#ifdef UNDEFINED

//class SampleFile : public Abst_SoundFile {
//public:
//  SampleFile();                           /// Constructor
//  SampleFile(string name, Abst_SoundFile *file = 0, int start = 1, int stop = -1);
//  ~SampleFile();
//                          /// Data members
//  unsigned mMIDIKey;          // sample's MIDI key #
//  double mFrequency;          // sample's actual frequency
//  double mMinRatio;           // min transp.
//  double mMaxRatio;           // max transp. (often 1.0)
//  
//  double ratioForKey(int desiredMIDI);
//  double ratioForPitch(int desiredMIDI);

SampleFile::SampleFile() {

}

SampleFile::SampleFile(string tpath, unsigned MIDIKey, double frequency, double minRatio, double maxRatio) {

}

SampleFile::~SampleFile() {

}

#endif