Mercurial > octave-nkf
diff libinterp/dldfcn/audioread.cc @ 19544:99522db5b911
merge audio source files
* audiodevinfo.cc: Fold __player_audioplayer__.cc,
__recorder_audiorecorder__.cc, player_class.cc, player_class.h,
recorder_class.cc, and recorder_class.h into this source file.
* audioread.cc: Fold audioinfo.cc and audiowrite.cc into this source
file.
* libinterp/dldfcn/module-files: Update.
author | John W. Eaton <jwe@octave.org> |
---|---|
date | Fri, 02 Jan 2015 00:40:35 -0500 |
parents | ce02743b6f2a |
children | 19f75d156ffe |
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line diff
--- a/libinterp/dldfcn/audioread.cc Wed Dec 31 15:38:13 2014 -0500 +++ b/libinterp/dldfcn/audioread.cc Fri Jan 02 00:40:35 2015 -0500 @@ -24,8 +24,12 @@ #include <config.h> #endif +#include <string> +#include <map> + #include "oct.h" #include "ov-struct.h" + #ifdef HAVE_SNDFILE #include <sndfile.h> #endif @@ -126,3 +130,191 @@ #endif return octave_value (retval); } + +#ifdef HAVE_SNDFILE +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; +} +#endif + +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},\n\ +file format will be determined by the file extension.\n\ +\n\ +@end deftypefn\n\ +@deftypefn {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\n\ +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") +{ + octave_scalar_map retval; +#ifdef HAVE_SNDFILE + 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); + 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); +#else + error ("sndfile not found on your system and thus audiowrite is not functional"); +#endif + return octave_value (retval); +} + +DEFUN_DLD (audioinfo, args, , + "-*- texinfo -*-\n\ +@deftypefn {Loadable Function} {@var{info} =} audioinfo (@var{filename})\n\ +Return information about an audio file specified by @var{filename}.\n\ +@end deftypefn") +{ + octave_scalar_map retval; + if (args.length () != 1 || not args(0).is_string ()) + { + print_usage (); + return octave_value (retval); + } +#ifdef HAVE_SNDFILE + Matrix audio; + SNDFILE *file; + SF_INFO info; + info.format = 0; + int start, end; + file = sf_open (args(0).string_value ().c_str (), SFM_READ, &info); + retval.assign ("Filename", args(0).string_value ()); + retval.assign ("CompressionMethod", ""); + retval.assign ("NumChannels", info.channels); + retval.assign ("SampleRate", info.samplerate); + retval.assign ("TotalSamples", info.frames); + retval.assign ("Duration", (float)info.frames / (float)info.samplerate); + + int bits; + if (info.format & SF_FORMAT_PCM_S8) + bits = 8; + else if (info.format & SF_FORMAT_PCM_U8) + bits = 8; + else if (info.format & SF_FORMAT_PCM_16) + bits = 16; + else if (info.format & SF_FORMAT_PCM_24) + bits = 24; + else if (info.format & SF_FORMAT_PCM_32) + bits = 32; + else + bits = -1; + + retval.assign ("BitsPerSample", bits); + retval.assign ("BitRate", -1); + retval.assign ("Title", sf_get_string (file, SF_STR_TITLE)); + retval.assign ("Artist", sf_get_string (file, SF_STR_ARTIST)); + retval.assign ("Comment", sf_get_string (file, SF_STR_COMMENT)); +#else + error ("sndfile not found on your system and thus audioinfo is not functional"); +#endif + return octave_value (retval); +}