# HG changeset patch # User Vytautas Jančauskas # Date 1378803055 -10800 # Node ID 0243b86b3cee8d7b4753642dd911750bc178ae98 # Parent 5eae41c8c07f54eb26c4f814bf61de287ff5e5aa Added libsndfile based audio file utilities to the source tree * audioinfo.cc, audioread.cc, audiowrite.cc: new files * module-files: appended with lines to build new files diff -r 5eae41c8c07f -r 0243b86b3cee libinterp/dldfcn/audioinfo.cc --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libinterp/dldfcn/audioinfo.cc Tue Sep 10 11:50:55 2013 +0300 @@ -0,0 +1,58 @@ +#include +#include +#include + +DEFUN_DLD(audioinfo, args, , +"-*- texinfo -*-\n\ +@deftypefn{Loadable Function} info = audioinfo(@var{filename})\n\ +\n\ +Return information about an audio file specified by @var{filename}.\ +\n\ +@end deftypefn" +) +{ + octave_scalar_map retval; + 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)); + return octave_value(retval); +} diff -r 5eae41c8c07f -r 0243b86b3cee libinterp/dldfcn/audioread.cc --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libinterp/dldfcn/audioread.cc Tue Sep 10 11:50:55 2013 +0300 @@ -0,0 +1,109 @@ +#include +#include +#include + +DEFUN_DLD(audioread, args, , +"-*- texinfo -*-\n\ +@deftypefn{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename})\n\ +\n\ +Load an audio file that is specified by @var{filename}. It will be loaded in to \ +a column matrix with as many rows as there are audio frames and as many columns \ +as there are channels in the file. Sampling rate will be stored in @var{Fs}. \ +\n\ +@deftypefnx{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename}, @var{samples})\n\ +\n\ +Read a specified range of samples from a file specified by @var{filename}. \ +Argument @var{samples} is a vector with two values specifying starting frame \ +and ending frame. \ +\n\ +@deftypefnx{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename}, @var{dataType})\n\ +\n\ +Read a file and return an array of specified type. If @var{dataType} is \"native\" then \ +an array of fixed width integer type will be returned depending on how data is stored \ +in the audio file. If @var{dataType} is \"double\" a double matrix will be returned. \ +\n\ +@deftypefnx{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename}, @var{samples}, @var{dataType})\n\ +\n\ +Read a file and return a specified range of frames in an array of specified type. \ +\n\ +@end deftypefn" +) +{ + octave_value_list retval; + Matrix audio; + octave_value ret_audio; + SNDFILE *file; + SF_INFO info; + info.format = 0; + int start, end; + file = sf_open(args(0).string_value ().c_str (), SFM_READ, &info); + start = 0; + end = info.frames; + float *data = (float *)malloc (sizeof(float) * info.frames * info.channels); + sf_read_float(file, data, info.frames * info.channels); + if (args.length () == 2 && !args(1).is_string () || args.length () == 3) + { + RowVector range = args(1).row_vector_value (); + start = range(0); + end = range(1); + } + audio.resize (end - start, info.channels); + for (int i = start; i < end; i++) + { + for (int channel = 0; channel < info.channels; channel++) + { + audio(i - start, channel) = data[i * info.channels + channel]; + } + } + free (data); + if (args.length () == 2 && args(1).is_string () || args.length () == 3) + { + std::string type; + if (args.length () == 3) + { + type = args(2).string_value (); + } + else + { + type = args(1).string_value (); + } + if (type == "native") + { + if (info.format & SF_FORMAT_PCM_S8) + { + ret_audio = octave_value ((audio * 127)).int8_array_value (); + } + else if (info.format & SF_FORMAT_PCM_U8) + { + ret_audio = octave_value ((audio * 127 + 127)).uint8_array_value (); + } + else if (info.format & SF_FORMAT_PCM_16) + { + ret_audio = octave_value ((audio * 32767)).int16_array_value (); + } + else if (info.format & SF_FORMAT_PCM_24) + { + ret_audio = octave_value ((audio * 8388608)).int32_array_value (); + } + else if (info.format & SF_FORMAT_PCM_32) + { + ret_audio = octave_value ((audio * 2147483648)).int32_array_value (); + } + else + { + ret_audio = octave_value (audio); + } + } + else + { + ret_audio = octave_value (audio); + } + } + else + { + ret_audio = octave_value (audio); + } + retval(0) = ret_audio; + retval(1) = info.samplerate; + return octave_value(retval); +} diff -r 5eae41c8c07f -r 0243b86b3cee libinterp/dldfcn/audiowrite.cc --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libinterp/dldfcn/audiowrite.cc Tue Sep 10 11:50:55 2013 +0300 @@ -0,0 +1,164 @@ +#include +#include +#include +#include +#include + +static void fill_extension_table(std::map &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 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); +} diff -r 5eae41c8c07f -r 0243b86b3cee libinterp/dldfcn/module-files --- a/libinterp/dldfcn/module-files Tue Oct 01 18:02:23 2013 -0700 +++ b/libinterp/dldfcn/module-files Tue Sep 10 11:50:55 2013 +0300 @@ -20,3 +20,6 @@ symrcm.cc|$(SPARSE_XCPPFLAGS)|$(SPARSE_XLDFLAGS)|$(SPARSE_XLIBS) tsearch.cc urlwrite.cc|$(CURL_CPPFLAGS)|$(CURL_LDFLAGS)|$(CURL_LIBS) +audioread.cc|-lsndfile +audiowrite.cc|-lsndfile +audioinfo.cc|-lsndfile \ No newline at end of file