// File: savewav.cc
// Description: Save Octave matrix in WAV file 
// Author: Rafael Laboissière <address@hidden>
// Created on: Sat Jan 23 21:57:00 CET 1999
// Last modified on: Mon Jan 25 23:40:00 CET 1999
// $Id: savewav.cc 1.2 Mon, 25 Jan 1999 23:40:04 +0100 rafael $

// Copyright (C) 1999 Rafael Laboissiere
// This file is free software distributed under the GNU Public License 
// No warranty.  Use it at your own risk.

#include <oct.h>
#include <sys/types.h>
#include <math.h>
#include <audiofile.h>

DEFUN_DLD( savewav, args, nargout, "\
SYNOPSIS:\n\
\n\
    savewav(\"filename.wav\", Sample_Rate, Sample_Width, Data )\n\
\n\
DESCRIPTION:\n\
\n\
  Saves matrix Data in the M$ WAV file \"filename.wav\".  The number\n\
  of rows and columns of matrix Data are equal to the number of samples\n\
  and the number of channels of the WAV file, respectively.  Sample_Rate is\n\
  the number of samples per second, and Sample_width the number of bits of\n\
  each sample.\n\
\n\
  The present version of savewav can only write files whose Sample_Width\n\
  are 8, 16 or 32 bits.  In conformance with the WAV format definition,\n\
  when Sample_Width is equal to 8 the samples are unsigned\n\
  (values between 0 and 255).  When Sample_width is greater than 8, the\n\
  samples are signed." )
{
  AFfilehandle file;
  AFframecount frameCount;
  AFfilesetup outputSetup;
  int sampleFormat, sampleWidth, channelCount;
  double sampleRate;
  octave_value_list retval;

  // Check if argument is string
  if ( !args(0).is_string() ) {
    error("savewav: Arg must be a string!\n");
    return retval;
  }

  // Convert argument into an ordinary (char *), needed for afOpenFile
  int filename_len = args(0).string_value().size();
  char* filename = new char[filename_len + 1];
  args(0).string_value().copy(filename, filename_len);
  filename[filename_len] = 0;

  // Set sample rate 
  if ( ! args(1).is_real_scalar() ) {
    error("savewav: Sample Rate should be a real scalar.\n");
    return retval;
  }
  sampleRate =  args(1).double_value();

  // Set sample width & sample format
  if ( ! args(2).is_real_scalar() ) {
    error("savewav: Sample width should be a real scalar.\n");
    return retval;
  }
  sampleWidth = (int) args(2).double_value();
  if ( sampleWidth == 8 )
    sampleFormat = AF_SAMPFMT_UNSIGNED;
  else
    if ( (sampleWidth == 16) || (sampleWidth == 32) )
    sampleFormat = AF_SAMPFMT_TWOSCOMP;
    else {
      error("savewav: Sample width should be equal to 8, 16 or 32.\n");
      return retval;
  }

  // Get data, as well as frame and channel counts
  Matrix data = args(3).matrix_value();
  frameCount =  data.rows();
  channelCount = data.cols();

  // Transfer samples
  char* buffer = new char[ frameCount * channelCount * (sampleWidth/8) ];
  u_int8_t* int8_p =  (u_int8_t *) buffer;
  int16_t* int16_p =  (int16_t *) buffer;
  int32_t* int32_p =  (int32_t *) buffer;
  for (int i = 0; i < frameCount; i++)
    for (int j = 0; j < channelCount; j++)
      if ( sampleWidth == 8 )
	*(int8_p++) = (u_int8_t) data(i,j);
      else 
	if ( sampleWidth == 16 )
	  *(int16_p++) = (int16_t) data(i,j);
	else
	  *(int32_p++) = (int32_t) data(i,j);

  // Initialize output file setup
  outputSetup = afNewFileSetup();
  afInitFileFormat(outputSetup, AF_FILE_WAVE);
  afInitRate(outputSetup, AF_DEFAULT_TRACK, sampleRate);
  afInitChannels(outputSetup, AF_DEFAULT_TRACK, channelCount);
  afInitSampleFormat (outputSetup, AF_DEFAULT_TRACK, sampleFormat,
		      sampleWidth);
  afInitByteOrder (outputSetup, AF_DEFAULT_TRACK, AF_BYTEORDER_LITTLEENDIAN);
  // Open WAV file and check for success
  file = afOpenFile(filename, "w", outputSetup);
  if (file == NULL) {
    error("savewav: Unable to open output file.\n");
    return retval;
  }
  afFreeFileSetup(outputSetup);
  // Write samples
  afWriteFrames(file, AF_DEFAULT_TRACK, buffer, frameCount);
  afCloseFile(file);

  return retval;
}