audioread.cc

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////////////////////////////////////////////////////////////////////////
//
// Copyright (C) 2013-2025 The Octave Project Developers
//
// See the file COPYRIGHT.md in the top-level directory of this
// distribution or <https://octave.org/copyright/>.
//
// This file is part of Octave.
//
// Octave 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 3 of the License, or
// (at your option) any later version.
//
// Octave 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 Octave; see the file COPYING.  If not, see
// <https://www.gnu.org/licenses/>.
//
////////////////////////////////////////////////////////////////////////
 
#if defined (HAVE_CONFIG_H)
#  include "config.h"
#endif
 
#include <algorithm>
#include <map>
#include <string>
 
#include "dMatrix.h"
#include "dRowVector.h"
#include "file-ops.h"
#include "file-stat.h"
#include "oct-locbuf.h"
#include "unwind-prot.h"
 
#include "defun-dld.h"
#include "error.h"
#include "errwarn.h"
#include "ov.h"
#include "ovl.h"
#include "pager.h"
 
#if defined (HAVE_SNDFILE)
#  include <sndfile.h>
#endif
 
OCTAVE_BEGIN_NAMESPACE(octave)
 
DEFUN_DLD (audioread, args, ,
           doc: /* -*- texinfo -*-
@deftypefn  {} {[@var{y}, @var{fs}] =} audioread (@var{filename})
@deftypefnx {} {[@var{y}, @var{fs}] =} audioread (@var{filename}, @var{samples})
 
@deftypefnx {} {[@var{y}, @var{fs}] =} audioread (@var{filename}, @var{datatype})
@deftypefnx {} {[@var{y}, @var{fs}] =} audioread (@var{filename}, @var{samples}, @var{datatype})
Read the audio file @var{filename} and return the audio data @var{y} and
sampling rate @var{fs}.
 
The audio data is stored as matrix with rows corresponding to audio frames
and columns corresponding to channels.
 
The optional two-element vector argument @var{samples} specifies starting
and ending frames.
 
The optional argument @var{datatype} specifies the datatype to return.
If it is @qcode{"native"}, then the type of data depends on how the data
is stored in the audio file.
@seealso{audiowrite, audioformats, audioinfo}
@end deftypefn */)
{
#if defined (HAVE_SNDFILE)
 
  int nargin = args.length ();
 
  if (nargin < 1 || nargin > 3)
    print_usage ();
 
  std::string filename = args(0).xstring_value ("audioread: FILENAME must be a string");
 
  SF_INFO info;
  info.format = 0;
  SNDFILE *file = sf_open (filename.c_str (), SFM_READ, &info);
 
  if (! file)
    error ("audioread: failed to open input file '%s': %s",
           filename.c_str (), sf_strerror (file));
 
  unwind_action close_open_file ([file] () { sf_close (file); });
 
  // FIXME: It would be nicer to use a C++ expandable data container and
  // read a file of unknown length into memory in chunks and determine the
  // number of samples after reading.  See bug #60888.
  if (info.frames == SF_COUNT_MAX)
    error ("audioread: malformed header does not specify number of samples");
 
  OCTAVE_LOCAL_BUFFER (double, data, info.frames * info.channels);
 
  sf_read_double (file, data, info.frames * info.channels);
 
  sf_count_t start = 0;
  sf_count_t end = info.frames;
 
  if ((nargin == 2 && ! args(1).is_string ()) || nargin == 3)
    {
      RowVector range = args(1).row_vector_value ();
 
      if (range.numel () != 2)
        error ("audioread: invalid specification for range of frames");
 
      double dstart = (math::isinf (range(0)) ? info.frames : range(0));
      double dend = (math::isinf (range(1)) ? info.frames : range(1));
 
      if (dstart < 1 || dstart > dend || dend > info.frames
          || math::x_nint (dstart) != dstart
          || math::x_nint (dend) != dend)
        error ("audioread: invalid specification for range of frames");
 
      start = dstart - 1;
      end = dend;
    }
 
  sf_count_t items = end - start;
 
  Matrix audio (items, info.channels);
 
  double *paudio = audio.rwdata ();
 
  data += start * info.channels;
 
  for (int i = 0; i < items; i++)
    {
      for (int channel = 0; channel < info.channels; channel++)
        paudio[items*channel+i] = *data++;
    }
 
  octave_value ret_audio;
 
  if ((nargin == 2 && args(1).is_string ()) || nargin == 3)
    {
      std::string type;
      if (nargin == 3)
        type = args(2).string_value ();
      else
        type = args(1).string_value ();
 
      if (type == "native")
        {
          switch (info.format & SF_FORMAT_SUBMASK)
            {
            case SF_FORMAT_PCM_S8:
              ret_audio = int8NDArray (audio * 128);
              break;
            case SF_FORMAT_PCM_U8:
              ret_audio = uint8NDArray (audio * 128 + 128);
              break;
            case SF_FORMAT_PCM_16:
              ret_audio = int16NDArray (audio * 32768);
              break;
            case SF_FORMAT_PCM_24:
              ret_audio = int32NDArray (audio * 8388608);
              break;
            case SF_FORMAT_PCM_32:
              ret_audio = int32NDArray (audio * 2147483648);
              break;
            case SF_FORMAT_FLOAT:
              ret_audio = FloatNDArray (audio);
              break;
            default:
              ret_audio = audio;
              break;
            }
        }
      else
        ret_audio = audio;
    }
  else
    ret_audio = audio;
 
  return ovl (ret_audio, info.samplerate);
 
#else
 
  octave_unused_parameter (args);
 
  err_disabled_feature ("audioread",
                        "reading and writing sound files through libsndfile");
 
#endif
}
 
#if defined (HAVE_SNDFILE)
 
static int
extension_to_format (const std::string& ext)
{
  static bool initialized = false;
 
  static std::map<std::string, int> table;
 
  if (! initialized)
    {
      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;
#if defined (HAVE_LIB_SNDFILE_FORMAT_MP3)
      table["m1a"] = SF_FORMAT_MPEG;
#endif
 
      initialized = true;
    }
 
  std::map<std::string, int>::const_iterator it = table.find (ext);
 
  return (it != table.end ()) ? it->second : 0;
}
 
#endif
 
DEFUN_DLD (audiowrite, args, ,
           doc: /* -*- texinfo -*-
@deftypefn  {} {} audiowrite (@var{filename}, @var{y}, @var{fs})
@deftypefnx {} {} audiowrite (@var{filename}, @var{y}, @var{fs}, @var{name}, @var{value}, @dots{})
 
Write audio data from the matrix @var{y} to @var{filename} at sampling rate
@var{fs} with the file format determined by the file extension.
 
Additional name/value argument pairs may be used to specify the
following options:
 
@table @samp
@item BitsPerSample
Number of bits per sample.  Valid values are 8, 16, 24, and 32.  Default is
16.
 
@item BitRate
Valid argument name, but ignored.  Left for compatibility with @sc{matlab}.
 
@item Quality
Quality setting for the Ogg Vorbis compressor.  Values can range between 0
and 100 with 100 being the highest quality setting.  Default is 75.
 
@item Title
Title for the audio file.
 
@item Artist
Artist name.
 
@item Comment
Comment.
@end table
@seealso{audioread, audioformats, audioinfo}
@end deftypefn */)
{
#if defined (HAVE_SNDFILE)
 
  int nargin = args.length ();
 
  if (nargin < 3)
    print_usage ();
 
  std::string filename = args(0).xstring_value ("audiowrite: FILENAME must be a string");
 
  double bias = 0.0;
  double scale = 1.0;
 
  if (args(1).is_uint8_type ())
    bias = scale = 127.5;
  else if (args(1).is_int16_type ())
    scale = 32768;       // 2^15
  else if (args(1).is_int32_type ())
    scale = 2147483648;  // 2^31
  else if (args(1).isinteger ())
    err_wrong_type_arg ("audiowrite", args(1));
 
  Matrix audio = args(1).matrix_value ();
 
  if (! args(2).is_scalar_type () || ! args(2).isnumeric ())
    error ("audiowrite: sample rate FS must be a positive scalar integer");
  int samplerate = args(2).int_value ();
  if (samplerate < 1)
    error ("audiowrite: sample rate FS must be a positive scalar integer");
 
  std::string ext;
  std::size_t dotpos = filename.find_last_of ('.');
  if (dotpos != std::string::npos)
    ext = filename.substr (dotpos + 1);
  std::transform (ext.begin (), ext.end (), ext.begin (), ::tolower);
 
  sf_count_t items_to_write = audio.rows () * audio.columns ();
 
  if (audio.rows () == 1)
    audio = audio.transpose ();
 
  OCTAVE_LOCAL_BUFFER (double, data, items_to_write);
 
  sf_count_t idx = 0;
  for (int i = 0; i < audio.rows (); i++)
    {
      for (int j = 0; j < audio.columns (); j++)
        {
          double elem = (audio.xelem (i, j) - bias) / scale;
          data[idx++] = std::min (std::max (elem, -1.0), 1.0);
        }
    }
 
  SF_INFO info;
 
  memset (&info, 0, sizeof (info));
 
  sf_count_t chunk_size = 0;
 
  if (ext == "ogg")
    {
      info.format = SF_FORMAT_VORBIS;
 
      // FIXME: There seems to be a bug writing ogg files in one shot that
      // causes a segfault: https://bugs.debian.org/760898.
      // Breaking it up into a series of smaller chunks appears to avoid the
      // problem and produces valid files.
      chunk_size = 0x100000;
    }
#if defined (HAVE_LIB_SNDFILE_FORMAT_MP3)
  else if (ext == "mp1")
    info.format = SF_FORMAT_MPEG|SF_FORMAT_MPEG_LAYER_I;
  else if (ext == "mp2")
    info.format = SF_FORMAT_MPEG|SF_FORMAT_MPEG_LAYER_II;
  else if (ext == "mp3")
    info.format = SF_FORMAT_MPEG|SF_FORMAT_MPEG_LAYER_III;
#endif
  else
    info.format = SF_FORMAT_PCM_16;
 
  info.channels = audio.columns ();
  info.samplerate = samplerate;
  info.format |= extension_to_format (ext);
 
  std::string title = "";
  std::string artist = "";
  std::string comment = "";
  double quality = 0.75;
 
  for (int i = 3; i < nargin; i += 2)
    {
      if (i >= nargin - 1)
        error ("audiowrite: invalid number of arguments");
 
      std::string keyword_orig = args(i).string_value ();
      std::string keyword = args(i).xtolower ().string_value ();
      octave_value value_arg = args(i+1);
 
      if (keyword == "bitspersample")
        {
          info.format &= ~SF_FORMAT_SUBMASK;
          int bits = value_arg.int_value ();
          if (bits == 8)
            {
              if ((info.format & SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV)
                info.format |= SF_FORMAT_PCM_U8;
              else
                info.format |= SF_FORMAT_PCM_S8;
            }
          else if (bits == 16)
            info.format |= SF_FORMAT_PCM_16;
          else if (bits == 24)
            {
              if ((info.format & SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV)
                info.format |= SF_FORMAT_PCM_32;
              else
                info.format |= SF_FORMAT_PCM_24;
            }
          else if (bits == 32)
            {
              if ((info.format & SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
                  && args(1).isfloat ())
                info.format |= SF_FORMAT_FLOAT;
              else
                info.format |= SF_FORMAT_PCM_32;
            }
          else if (bits == 64)
            info.format |= SF_FORMAT_DOUBLE;
          else
            error ("audiowrite: wrong number of bits specified");
        }
      else if (keyword == "bitrate")
        warning_with_id ("Octave:audiowrite:unused-parameter",
                         "audiowrite: 'BitRate' accepted for Matlab "
                         "compatibility, but is ignored");
      else if (keyword == "quality")
        {
          if (! value_arg.is_scalar_type ())
            error ("audiowrite: Quality value must be a scalar");
 
          double value =
            value_arg.xdouble_value ("audiowrite: Quality value must be a numeric scalar between 0 and 100");
 
          if (math::isnan (value) || value < 0 || value > 100)
            error ("audiowrite: Quality value must be a number between 0 and 100");
 
          quality = value / 100;
        }
      else if (keyword == "title")
        title = value_arg.string_value ();
      else if (keyword == "artist")
        artist = value_arg.string_value ();
      else if (keyword == "comment")
        comment = value_arg.string_value ();
      else
        error ("audiowrite: unrecognized option: '%s'", keyword_orig.c_str ());
    }
 
  SNDFILE *file = sf_open (filename.c_str (), SFM_WRITE, &info);
 
  if (! file)
    error ("audiowrite: failed to open output file '%s': %s",
           filename.c_str (), sf_strerror (file));
 
  unwind_action close_open_file ([file] () { sf_close (file); });
 
  sf_command (file, SFC_SET_NORM_DOUBLE, nullptr, SF_TRUE);
  sf_command (file, SFC_SET_CLIPPING, nullptr, SF_TRUE) ;
  sf_command (file, SFC_SET_VBR_ENCODING_QUALITY, &quality, sizeof (quality));
 
  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_count_t total_items_written = 0;
  sf_count_t offset = 0;
 
  if (chunk_size == 0)
    chunk_size = items_to_write;
 
  while (total_items_written < items_to_write)
    {
      if (items_to_write - offset < chunk_size)
        chunk_size = items_to_write - offset;
 
      sf_count_t items_written = sf_write_double (file, data+offset, chunk_size);
 
      if (items_written != chunk_size)
        error ("audiowrite: write failed, wrote %" PRId64 " of %" PRId64
               " items\n", items_written, chunk_size);
 
      total_items_written += items_written;
      offset += chunk_size;
    }
 
  // FIXME: Shouldn't we return something to indicate whether the file
  //        was written successfully?  On the other hand, Matlab doesn't
  //        return anything.
  return ovl ();
 
#else
 
  octave_unused_parameter (args);
 
  err_disabled_feature ("audiowrite",
                        "reading and writing sound files through libsndfile");
 
#endif
}
 
/*
## Joint audiowrite/audioread tests
## 8-bit Unsigned PCM
%!testif HAVE_SNDFILE <*56889>
%! fname = [tempname() ".wav"];
%! unwind_protect
%!   y1 = uint8 ([0, 1, 2, 253, 254, 255]);
%!   audiowrite (fname, y1, 8000, "BitsPerSample", 8);
%!   y2 = audioread (fname, "native");
%! unwind_protect_cleanup
%!   unlink (fname);
%! end_unwind_protect
%! assert (y1(:), y2);
 
## 8-bit Signed PCM
%!testif HAVE_SNDFILE <*56889>
%! fname = [tempname() ".au"];
%! unwind_protect
%!   y1 = uint8 ([0, 1, 2, 253, 254, 255]);
%!   audiowrite (fname, y1, 8000, "BitsPerSample", 8);
%!   y2 = audioread (fname, "native");
%! unwind_protect_cleanup
%!   unlink (fname);
%! end_unwind_protect
%! assert (y2, int8 ([-128; -127; -126; 125; 126; 127]));
 
## 16-bit Signed PCM
%!testif HAVE_SNDFILE <*56889>
%! fname = [tempname() ".wav"];
%! unwind_protect
%!   y1 = int16 ([-32768, -32767, -32766, 32765, 32766, 32767]);
%!   audiowrite (fname, y1, 8000, "BitsPerSample", 16);
%!   y2 = audioread (fname, "native");
%! unwind_protect_cleanup
%!   unlink (fname);
%! end_unwind_protect
%! assert (y1(:), y2);
 
## 24-bit Signed PCM
%!testif HAVE_SNDFILE <*56889>
%! fname = [tempname() ".au"];
%! unwind_protect
%!   y1 = [-8388608, -8388607, -8388606, 8388605, 8388606, 8388607] / 8388608;
%!   audiowrite (fname, y1, 8000, "BitsPerSample", 24);
%!   y2 = audioread (fname, "native");
%! unwind_protect_cleanup
%!   unlink (fname);
%! end_unwind_protect
%! assert (int32 ([-8388608; -8388607; -8388606; 8388605; 8388606; 8388607]),
%!         y2);
 
## 32-bit Signed PCM
%!testif HAVE_SNDFILE <*56889>
%! fname = [tempname() ".wav"];
%! unwind_protect
%!   y1 = int32 ([-2147483648, -2147483647, -2147483646, 2147483645, 2147483646, 2147483647 ]);
%!   audiowrite (fname, y1, 8000, "BitsPerSample", 32);
%!   y2 = audioread (fname, "native");
%! unwind_protect_cleanup
%!   unlink (fname);
%! end_unwind_protect
%! assert (y1(:), y2);
 
## Test input validation
%!testif HAVE_SNDFILE
%! fail ("audiowrite (1, 1, 8e3)", "FILENAME must be a string");
%! fail ("audiowrite ('foo', int64 (1), 8e3)",
%!       "wrong type argument 'int64 scalar'");
%! fail ("audiowrite ('foo', [0 1], [8e3, 8e3])",
%!       "FS must be a positive scalar");
%! fail ("audiowrite ('foo', 1, {8e3})", "FS must be a .* integer");
%! fail ("audiowrite ('foo', 1, -8e3)", "FS must be a positive");
%! fail ("audiowrite ('foo', 1, 8e3, 'bitspersample')",
%!       "invalid number of arguments");
%! fail ("audiowrite ('foo', 1, 8e3, 'bitspersample', 48)",
%!       "wrong number of bits specified");
%! fail ("audiowrite ('foo', 1, 8e3, 'quality', [2 3 4])",
%!       "Quality value must be a scalar");
%! fail ("audiowrite ('foo', 1, 8e3, 'quality', NaN)",
%!       "Quality value must be .* between 0 and 100");
%! fail ("audiowrite ('foo', 1, 8e3, 'quality', -1)",
%!       "Quality value must be .* between 0 and 100");
%! fail ("audiowrite ('foo', 1, 8e3, 'quality', 101)",
%!       "Quality value must be .* between 0 and 100");
%! fail ("audiowrite ('foo', 1, 8e3, 'foo', 'bar')",
%!       "unrecognized option: 'foo'");
*/
 
DEFUN_DLD (audioinfo, args, ,
           doc: /* -*- texinfo -*-
@deftypefn {} {@var{info} =} audioinfo (@var{filename})
Return information about an audio file specified by @var{filename}.
 
The output @var{info} is a structure containing the following fields:
 
@table @samp
@item Filename
Name of the audio file.
 
@item CompressionMethod
Audio compression method.  Unused, only present for compatibility with
@sc{matlab}.
 
@item NumChannels
Number of audio channels.
 
@item SampleRate
Sample rate of the audio, in Hertz.
 
@item TotalSamples
Number of samples in the file.
 
@item Duration
Duration of the audio, in seconds.
 
@item BitsPerSample
Number of bits per sample.
 
@item BitRate
Audio bit rate.  Unused, only present for compatibility with @sc{matlab}.
 
@item Title
@qcode{"Title"} audio metadata value as a string, or empty if not present.
 
@item Artist
@qcode{"Artist"} audio metadata value as a string, or empty if not present.
 
@item Comment
@qcode{"Comment"} audio metadata value as a string, or empty if not present.
@end table
@seealso{audioread, audiowrite}
@end deftypefn */)
{
#if defined (HAVE_SNDFILE)
 
  if (args.length () != 1)
    print_usage ();
 
  std::string filename = args(0).xstring_value ("audioinfo: FILENAME must be a string");
 
  sys::file_stat fs (filename);
  if (! fs.exists ())
    error ("audioinfo: FILENAME '%s' not found", filename.c_str ());
 
  SF_INFO info;
  info.format = 0;
  SNDFILE *file = sf_open (filename.c_str (), SFM_READ, &info);
 
  if (! file)
    error ("audioinfo: failed to open input file '%s': %s",
           filename.c_str (), sf_strerror (file));
 
  unwind_action close_open_file ([file] () { sf_close (file); });
 
  octave_scalar_map result;
 
  std::string full_name = sys::canonicalize_file_name (filename);
 
  result.assign ("Filename", full_name);
  result.assign ("CompressionMethod", "");
  result.assign ("NumChannels", info.channels);
  result.assign ("SampleRate", info.samplerate);
  double dframes;
  if (info.frames != SF_COUNT_MAX)
    dframes = info.frames;
  else
    dframes = -1;
  result.assign ("TotalSamples", dframes);
 
  if (dframes != -1)
    {
      double drate = info.samplerate;
      result.assign ("Duration", dframes / drate);
    }
  else
    result.assign ("Duration", -1);
 
  int bits;
  switch (info.format & SF_FORMAT_SUBMASK)
    {
    case SF_FORMAT_PCM_S8:
      bits = 8;
      break;
    case SF_FORMAT_PCM_U8:
      bits = 8;
      break;
    case SF_FORMAT_PCM_16:
      bits = 16;
      break;
    case SF_FORMAT_PCM_24:
      bits = 24;
      break;
    case SF_FORMAT_PCM_32:
      bits = 32;
      break;
    case SF_FORMAT_FLOAT:
      bits = 32;
      break;
    case SF_FORMAT_DOUBLE:
      bits = 64;
      break;
    default:
      bits = -1;
      break;
    }
 
  result.assign ("BitsPerSample", bits);
  result.assign ("BitRate", -1);
  result.assign ("Title", sf_get_string (file, SF_STR_TITLE));
  result.assign ("Artist", sf_get_string (file, SF_STR_ARTIST));
  result.assign ("Comment", sf_get_string (file, SF_STR_COMMENT));
 
  return ovl (result);
 
#else
 
  octave_unused_parameter (args);
 
  err_disabled_feature ("audioinfo",
                        "reading and writing sound files through libsndfile");
 
#endif
}
 
#if defined (HAVE_SNDFILE)
 
static void
audio_sub_formats (int format)
{
  int count;
  sf_command (nullptr, SFC_GET_FORMAT_SUBTYPE_COUNT, &count, sizeof (int));
 
  for (int i = 0; i < count; i++)
    {
      SF_FORMAT_INFO info;
      info.format = i;
      sf_command (nullptr, SFC_GET_FORMAT_SUBTYPE, &info, sizeof (info));
 
      SF_INFO sfinfo;
      memset (&sfinfo, 0, sizeof (sfinfo));
      sfinfo.channels = 1;
      sfinfo.format = (format & SF_FORMAT_TYPEMASK) | info.format;
 
      if (sf_format_check (&sfinfo))
        octave_stdout << "  " << info.name << std::endl;
    }
}
 
#endif
 
DEFUN_DLD (audioformats, args, ,
           doc: /* -*- texinfo -*-
@deftypefn  {} {} audioformats ()
@deftypefnx {} {} audioformats (@var{format})
Display information about all supported audio formats.
 
If the optional argument @var{format} is given, then display only formats
with names that start with @var{format}.
@seealso{audioread, audiowrite}
@end deftypefn */)
{
#if defined (HAVE_SNDFILE)
 
  if (args.length () > 1)
    print_usage ();
 
  std::string search = "";
  if (args.length () > 0)
    {
      search = args(0).string_value ();
      std::transform (search.begin (), search.end (), search.begin (), tolower);
    }
 
  int count;
  sf_command (nullptr, SFC_GET_FORMAT_MAJOR_COUNT, &count, sizeof (int));
 
  for (int i = 0; i < count; i++)
    {
      SF_FORMAT_INFO info;
      info.format = i;
      sf_command (nullptr, SFC_GET_FORMAT_MAJOR, &info, sizeof (info));
      bool match = true;
 
      if (! search.empty ())
        {
          std::string nm = info.name;
          std::transform (nm.begin (), nm.end (), nm.begin (), tolower);
          match = nm.compare (0, search.length (), search) == 0;
        }
 
      if (match)
        {
          octave_stdout << "name: " << info.name << std::endl;
          octave_stdout << "extension: " << info.extension << std::endl;
          octave_stdout << "id: " << info.format << std::endl;
          octave_stdout << "subformats:" << std::endl;
 
          audio_sub_formats (info.format);
        }
    }
 
  return octave_value_list ();
 
#else
 
  octave_unused_parameter (args);
 
  err_disabled_feature ("audioformats",
                        "getting sound formats through libsndfile");
 
#endif
}
 
OCTAVE_END_NAMESPACE(octave)