audiofile.cpp

sip 开源代码 源于novell sip 开源代码 源于novell

/**
 *  Copyright (C) 2005 Savoir-Faire Linux inc.
 *  Author: Yan Morin <[e-mail]>
 *
 *  Inspired by tonegenerator of 
 *   Laurielle Lea <[e-mail]> (2004)
 *  Inspired by ringbuffer of Audacity Project
 * 
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; either version 2 of the License, or
 *  (at your option) any later version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 */
#include "audiofile.h"
#include "codecDescriptor.h"
#include <fstream>
#include <math.h>
#include <samplerate.h>


AudioFile::AudioFile()
 : AudioLoop()
{
  // could vary later...
  _ulaw = new Ulaw(PAYLOAD_CODEC_ULAW);
  _start = false;
}


AudioFile::~AudioFile()
{
  delete _ulaw;
}

// load file in mono format
bool
AudioFile::loadFile(const std::string& filename, unsigned int sampleRate=8000) 
{
  // if the filename was already load, with the same samplerate 
  // we do nothing
  if (_filename == filename && _sampleRate == sampleRate) {
    return true;
  } else {
    // reset to 0
    delete [] _buffer; _buffer = 0;
    _size = 0;
    _pos = 0;
  }

  // no filename to load
  if (filename.empty()) {
    _debug("Unable to open audio file: filename is empty\n");
    return false;
  }

  std::fstream file;
  file.open(filename.c_str(), std::fstream::in);
  if (!file.is_open()) {
    // unable to load the file
    _debug("Unable to open audio file %s\n", filename.c_str());
    return false;
  }

  // get length of file:
  file.seekg (0, std::ios::end);
  int length = file.tellg();
  file.seekg (0, std::ios::beg);

  // allocate memory:
  char fileBuffer[length];

  // read data as a block:
  file.read (fileBuffer,length);
  file.close();

  // Decode file.ul
  // expandedsize is the number of bytes, not the number of int
  // expandedsize should be exactly two time more, else failed
  int16 monoBuffer[length];
  unsigned int expandedsize = _ulaw->codecDecode (monoBuffer, (unsigned char *) fileBuffer, length);
  if (expandedsize != length*2) {
    _debug("Audio file error on loading audio file!");
    return false;
  }
  unsigned int nbSampling = expandedsize/sizeof(int16);

  // we need to change the sample rating here:
  // case 1: we don't have to resample : only do splitting and convert
  if ( sampleRate == 8000 ) {
     // just s
     _size   = nbSampling;
     _buffer = new SFLDataFormat[_size];
#ifdef DATAFORMAT_IS_FLOAT
     // src to dest
     src_short_to_float_array(monoBuffer, _buffer, nbSampling);
#else
     // dest to src
     memcpy(_buffer, monoBuffer, _size*sizeof(SFLDataFormat));
#endif

  } else {
    // case 2: we need to convert it and split it
    // convert here
    double factord = (double)sampleRate / 8000;
    float* floatBufferIn = new float[nbSampling];
    int    sizeOut  = (int)(ceil(factord*nbSampling));
    src_short_to_float_array(monoBuffer, floatBufferIn, nbSampling);
    SFLDataFormat* bufferTmp = new SFLDataFormat[sizeOut];

    SRC_DATA src_data;
    src_data.data_in = floatBufferIn;
    src_data.input_frames = nbSampling;
    src_data.output_frames = sizeOut;
    src_data.src_ratio = factord;

#ifdef DATAFORMAT_IS_FLOAT
    // case number 2.1: the output is float32 : convert directly in _bufferTmp
    src_data.data_out = bufferTmp;
    src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
#else
   // case number 2.2: the output is int16 : convert and change to int16
   float* floatBufferOut = new float[sizeOut];
   src_data.data_out = floatBufferOut;

   src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
   src_float_to_short_array(floatBufferOut, bufferTmp, src_data.output_frames_gen);

   delete [] floatBufferOut;
#endif
   delete [] floatBufferIn;
   nbSampling = src_data.output_frames_gen; 

   // if we are in mono, we send the bufferTmp location and don't delete it
   // else we split the audio in 2 and put it into buffer
   _size = nbSampling;
   _buffer = bufferTmp;  // just send the buffer pointer;
   bufferTmp = 0;
  }

  return true;
}