soundeditor/qatsh/SampledSound.cpp

// SampledSound class implementation
//
//  A sound as an array of multi-channel samples
//
// QATSH Copyright 2009 Jean-Philippe MEURET <jpmeuret@free.fr>

#include "SampledSound.h"

#include <iostream>
#include <limits>
#include <cstring>

#include <sndfile.hh>


SampledSound::SampledSound(int nChannels, int nFrames, int nSamplingRate, 
			   double** adSamples)
: _nChannels(nChannels), _nFrames(nFrames), _nSamplingRate(nSamplingRate), _adSamples(0)
{
    if (_nChannels)
    {
	_adSamples = new double*[_nChannels];
	if (_nFrames && _adSamples)
	{
	    for (int nChanInd = 0; nChanInd < _nChannels; nChanInd++)
	    {
		_adSamples[nChanInd] = new double[_nFrames];
		if (adSamples && _adSamples[nChanInd])
		    memcpy(_adSamples[nChanInd], adSamples[nChanInd], nFrames*sizeof(double));
	    }
	}
    }
}

SampledSound::~SampledSound()
{
    freeSamples();
}

void SampledSound::freeSamples()
{
    if (_adSamples)
    {
	for (int nChanInd = 0; nChanInd < _nChannels; nChanInd++)
	    delete [] _adSamples[nChanInd];
	delete [] _adSamples;
	_adSamples = 0;
    }
}

int SampledSound::samplingRate() const
{
    return _nSamplingRate;
}

int SampledSound::nbChannels() const
{
    return _nChannels;
}

int SampledSound::nbFrames() const
{
    return _nFrames;
}


double** SampledSound::samples()
{
    return _adSamples;
}

double* SampledSound::channel(int nChanInd)
{
    return _adSamples[nChanInd];
}


bool SampledSound::load(const char* pszFileName)
{
    std::cout << "SampledSound : Loading sound file '" << pszFileName << "' ..." << std::endl;
    
    // Open sound file.
    SndfileHandle sndFileHdle(pszFileName, SFM_READ);

    // Check if all right till now.
    if (!sndFileHdle)
    {
	std::cerr << "SampledSound : Failed to open sound file '" << pszFileName 
		  << "' : " << sndFileHdle.strError() << std::endl;
	return false;
    }

    // Check for number of frames : as we load the whole sample into memory,
    // we support only up to numeric_limits<int>::max() frames (samples per channel).
    if (sndFileHdle.frames() > std::numeric_limits<int>::max())
    {
	std::cerr << "SampledSound : Sound file '" << pszFileName 
		  << "' too big (can't support more than " << std::numeric_limits<int>::max()
		  << " samples)" << std::endl;
	return false;
    }

    // Save sample properties
    _nFrames = (int)sndFileHdle.frames();
    _nChannels = sndFileHdle.channels();
    _nSamplingRate = sndFileHdle.samplerate();
    
    // Load the sample into memory.
    // 1) Allocate the sample arrays (1 per channel).
    freeSamples();
    _adSamples = new double*[_nChannels];
    for (int nChanInd = 0; nChanInd < _nChannels; nChanInd++)
	_adSamples[nChanInd] = new double[_nFrames];


    // 2) Allocate a buffer for IO
    const int knFramesToRead = 2048;
    double* aBuffer = new double[_nChannels*knFramesToRead];
    
    // 3) Read the file and store the samples into the arrays.
    sf_count_t nReadSamples = 0;
    sf_count_t nReadFrames;
    int nSampInd;
    int nCurrFrameInd = 0;
    while ((nReadFrames = sndFileHdle.readf(aBuffer, knFramesToRead)))
    {
	nSampInd = 0;
	while (nReadFrames--)
	{
	    for (int nChanInd = 0; nChanInd < _nChannels; nChanInd++)
		_adSamples[nChanInd][nCurrFrameInd] = aBuffer[nSampInd++];
	    nCurrFrameInd++;
	}
	nReadSamples += nSampInd;
    }
	delete [] aBuffer;

    std::cout << "... done (" << nReadSamples << " samples x channels read)" << std::endl;

    return nReadSamples == _nFrames * _nChannels;
}

//======================================================================
// File extension to libsndfile::SF_FORMAT_* major type conversion table
// Note: Was setup for libsndfile 1.0.17 (more formats may be supported now).
struct SSndFileMajorFormatDesc
{
    const char* pszFileExt;
    int nSndFileMajorFormat;
};

static SSndFileMajorFormatDesc KSndFileMajorType[] = 
{
    { ".wav",  SF_FORMAT_WAV   }, // Microsoft WAV format (little endian).
    { ".aif",  SF_FORMAT_AIFF  }, // Apple/SGI AIFF format (big endian).
    { ".aifc", SF_FORMAT_AIFF  }, // Apple/SGI AIFF format (big endian).
    { ".aiff", SF_FORMAT_AIFF  }, // Apple/SGI AIFF format (big endian).
    { ".au",   SF_FORMAT_AU    }, // Sun/NeXT AU format (big endian).
    { ".snd",  SF_FORMAT_AU    }, // Sun/NeXT AU format (big endian).
    { ".raw",  SF_FORMAT_RAW   }, // RAW PCM data.
    { ".pcm",  SF_FORMAT_RAW   }, // RAW PCM data.
    { ".voc",  SF_FORMAT_VOC   }, // VOC files.
    { ".caf",  SF_FORMAT_CAF   }, // Apple Core Audio File format
    { ".w64",  SF_FORMAT_W64   }, // Sonic Foundry's 64 bit RIFF/WAV
    { ".sf",   SF_FORMAT_IRCAM }, // Berkeley/IRCAM/CARL
    { ".paf",  SF_FORMAT_PAF   }, // Ensoniq PARIS file format.
    { ".iff",  SF_FORMAT_SVX   }, // Amiga IFF / SVX8 / SV16 format.
    { ".svx",  SF_FORMAT_SVX   }, // Amiga IFF / SVX8 / SV16 format.
    { ".nist", SF_FORMAT_NIST  }, // Sphere NIST format.
    { ".sph",  SF_FORMAT_NIST  }, // Sphere NIST format.
    { ".flac", SF_FORMAT_FLAC  }, // FLAC lossless file format
//    { ".", SF_FORMAT_MAT4 },  // Matlab (tm) V4.2 / GNU Octave 2.0
//    { ".", SF_FORMAT_MAT5 },  // Matlab (tm) V5.0 / GNU Octave 2.1
//    { ".", SF_FORMAT_PVF },   // Portable Voice Format
//    { ".", SF_FORMAT_XI },    // Fasttracker 2 Extended Instrument
//    { ".", SF_FORMAT_HTK },   // HMM Tool Kit format
//    { ".", SF_FORMAT_SDS },   // Midi Sample Dump Standard
//    { ".", SF_FORMAT_AVR },   // Audio Visual Research
//    { ".", SF_FORMAT_WAVEX }, // MS WAVE with WAVEFORMATEX
//    { ".", SF_FORMAT_SD2 },   // Sound Designer 2
    { 0, 0  } // End of data marker; don't remove.
};

// ESampleFormat to libsndfile::SF_FORMAT_* subtype conversion table
static int KSndFileSubType[SampledSound::eSampFormatNumber] =
{
    SF_FORMAT_PCM_16, // Signed 16 bit data
    SF_FORMAT_PCM_24, // Signed 24 bit data
    SF_FORMAT_PCM_32, // Signed 32 bit data
    
    SF_FORMAT_FLOAT,  // 32 bit float data
    SF_FORMAT_DOUBLE  // 64 bit float data
};

// Build a libsndfile format from the file extension (major format)
// and the sample format (format subtype) (see libsndfile API).
static int SndfileFormat(const char* pszFileExt, SampledSound::ESampleFormat eSampFormat)
{
    SSndFileMajorFormatDesc* pFormaDesc = KSndFileMajorType;
    while (pFormaDesc->pszFileExt && strcmp(pFormaDesc->pszFileExt, pszFileExt))
	pFormaDesc++;
    std::cout << "SampledSound : SndfileFormat(" << pszFileExt << ") : Major=" 
	      << std::hex << pFormaDesc->nSndFileMajorFormat 
	      << ", SubType=" << KSndFileSubType[eSampFormat] << std::dec << std::endl;
    return pFormaDesc->nSndFileMajorFormat | KSndFileSubType[eSampFormat];
}

// Store samples to the given file using given sample format.
bool SampledSound::store(const char* pszFileName, ESampleFormat eSampFormat) const
{
    std::cout << "SampledSound : Storing sound file '" << pszFileName << "' ..." << std::endl;

    // Determine sample output format and check if it is supported.
    const int format = SndfileFormat(strrchr(pszFileName, '.'), eSampFormat);
    if (!SndfileHandle::formatCheck(format, _nChannels, _nSamplingRate))
    {
	std::cerr << "SampledSound : Unsupported output format for '" << pszFileName 
		  << "'" << std::endl;
	return false;
    }
	
    // Try and open sound file.
    SndfileHandle sndFileHdle(pszFileName, SFM_WRITE, format, _nChannels, _nSamplingRate);
    if (!sndFileHdle)
    {
	std::cerr << "SampledSound : Failed to open sound file '" << pszFileName 
		  << "' : " << sndFileHdle.strError() << std::endl;
	return false;
    }

    // Write samples to file.
    const int nFrameBlockLen = 2048; // Nb of frames (1 frame = nbChannels() samples)
    double* aFrameBlock = new double[nFrameBlockLen*_nChannels];
    int nCurrFrameInd = 0; // Index in _adSamples[*] of the frame to write
    while (nCurrFrameInd < _nFrames) // For each possible frame block :
    {
	// Copy samples to the frame block
	int nSampInd = 0; // Index in aFrameBlock of the sample to copy from _adSamples
	while (nSampInd < nFrameBlockLen*_nChannels && nCurrFrameInd < _nFrames)
        {
	    for (int nChanInd = 0; nChanInd < _nChannels; nChanInd++)
		aFrameBlock[nSampInd++] = _adSamples[nChanInd][nCurrFrameInd];
	    nCurrFrameInd++;
	}

	// Write the frame block.
	if (sndFileHdle.writef(aFrameBlock, nSampInd / _nChannels) < nSampInd / _nChannels)
	{
	    std::cerr << "SampledSound : Failed to write frame block in '" << pszFileName 
		      << "' : " << sndFileHdle.strError() << std::endl;
	    return false;
	}
    }
    delete [] aFrameBlock;

    return true;
}