view libinterp/dldfcn/audiowrite.cc @ 19497:157eeaa53277

Fix includes in audioread, audiowrite and audioinfo * audioinfo.cc, audiowrite.cc, audioinfo.cc: changed octave header includes to use local headers
author Vytautas Jančauskas <unaudio@gmail.com>
date Tue, 10 Sep 2013 14:08:46 +0300
parents 0243b86b3cee
children 9161b7653392
line wrap: on
line source

#include "oct.h"
#include "ov-struct.h"
#include <sndfile.h>
#include <string>
#include <map>

static void fill_extension_table(std::map<std::string, int> &table)
{
  table["wav"] = SF_FORMAT_WAV;
  table["aiff"] = SF_FORMAT_AIFF;
  table["au"] = SF_FORMAT_AU;
  table["raw"] = SF_FORMAT_RAW;
  table["paf"] = SF_FORMAT_PAF;
  table["svx"] = SF_FORMAT_SVX;
  table["nist"] = SF_FORMAT_NIST;
  table["voc"] = SF_FORMAT_VOC;
  table["ircam"] = SF_FORMAT_IRCAM;
  table["w64"] = SF_FORMAT_W64;
  table["mat4"] = SF_FORMAT_MAT4;
  table["mat5"] = SF_FORMAT_MAT5;
  table["pvf"] = SF_FORMAT_PVF;
  table["xi"] = SF_FORMAT_XI;
  table["htk"] = SF_FORMAT_HTK;
  table["sds"] = SF_FORMAT_SDS;
  table["avr"] = SF_FORMAT_AVR;
  table["wavex"] = SF_FORMAT_WAVEX;
  table["sd2"] = SF_FORMAT_SD2;
  table["flac"] = SF_FORMAT_FLAC;
  table["caf"] = SF_FORMAT_CAF;
  table["wve"] = SF_FORMAT_WVE;
  table["ogg"] = SF_FORMAT_OGG;
  table["mpc2k"] = SF_FORMAT_MPC2K;
  table["rf64"] = SF_FORMAT_RF64;
}
  
DEFUN_DLD(audiowrite, args, ,
"-*- texinfo -*-\n\
@deftypefn{Loadable Function} audiowrite(@var{filename}, @var{y}, @var{Fs})\n\
\n\
Write audio data from the matrix @var{y} to a file specified by @var{filename}, \
file format will be determined by the file extension.\
\n\
@deftypefnx{Loadable Function} audiowrite(@var{filename}, @var{y}, @var{Fs}, @var{Name}, @var{Value})\n\
\n\
Lets you specify additional parameters when writing the file. Those parameters are given in the table below:\n\
\n\
@table @samp\n\
@item BitsPerSample\n\
Number of bits per sample, valid values are 8, 16, 24 and 32. Default is 16.\n\
@item BitRate\n\
Valid argument name, but ignored. Left for compatibility with MATLAB.\n\
@item Quality\n\
Quality setting for the Ogg Vorbis compressor. Values can range between 0 and 100 with 100 being the highest quality setting. Default is 75.\n\
@item Title\n\
Title for the audio file.\n\
@item Artist\n\
Artist name.\n\
@item Comment\n\
Comment.\n\
@end table\n\
@end deftypefn"
)
{
  std::map<std::string, int> extension_to_format;
  fill_extension_table(extension_to_format);
  std::string filename = args(0).string_value ();
  std::string extension = filename.substr(filename.find_last_of(".") + 1);
  std::transform(extension.begin(), extension.end(), extension.begin(), ::tolower);
  octave_scalar_map retval;
  Matrix audio = args(1).matrix_value ();
  SNDFILE *file;
  SF_INFO info;
  float *data = (float *)malloc(audio.rows () * audio.cols () * sizeof(float));
  for (int i = 0; i < audio.cols (); i++)
    {
      for (int j = 0; j < audio.rows (); j++)
        {
          data[j * audio.cols () + i] = audio(j, i);
        }
    }
  if (extension == "ogg")
    {
      info.format = SF_FORMAT_VORBIS;
    }
  else
    {
      info.format = SF_FORMAT_PCM_16;
    }
  std::string title = "";
  std::string artist = "";
  std::string comment = "";
  float quality = 0.75;
  for (int i = 3; i < args.length (); i += 2)
    {
      if (args(i).string_value () == "BitsPerSample")
        {
          int bits = args(i + 1).int_value ();
          if (bits == 8)
            {
              info.format |= SF_FORMAT_PCM_S8;
            }
          else if (bits == 16)
            {
              info.format |= SF_FORMAT_PCM_16;
            }
          else if (bits == 24)
            {
              info.format |= SF_FORMAT_PCM_24;
            }
          else if (bits == 32)
            {
              info.format |= SF_FORMAT_PCM_32;
            }
          else
            {
              error("audiowrite: wrong number of bits specified");
            }
        }
      else if (args(i).string_value () == "BitRate")
        {

        }
      else if (args(i).string_value () == "Quality")
        {
          quality = args(i + 1).int_value () * 0.01;
        }
      else if (args(i).string_value () == "Title")
        {
          title = args(i + 1).string_value ();
        }
      else if (args(i).string_value () == "Artist")
        {
          artist = args(i + 1).string_value ();
        }
      else if (args(i).string_value () == "Comment")
        {
          comment = args(i + 1).string_value ();
        }
      else
        {
          error("audiowrite: wrong argument name");
        }
    }
  info.samplerate = args(2).int_value ();
  info.channels = audio.cols ();
  info.format |= extension_to_format[extension];
  file = sf_open (filename.c_str (), SFM_WRITE, &info);
  if (title != "")
    {
      sf_set_string (file, SF_STR_TITLE, title.c_str ());
    }
  if (artist != "")
    {
      sf_set_string (file, SF_STR_ARTIST, artist.c_str ());
    }
  if (comment != "")
    {
      sf_set_string (file, SF_STR_COMMENT, comment.c_str ());
    }
  sf_write_float (file, data, audio.rows () * audio.cols ());
  sf_close (file);
  free (data);
  return octave_value(retval);
}