Mercurial > octave-nkf
diff libinterp/dldfcn/audioread.cc @ 19537:36a26a131209
Apply Octave coding style to audio project additions
* libinterp/dldfcn/__player_audioplayer__.cc,
libinterp/dldfcn/__recorder_audiorecorder__.cc,
libinterp/dldfcn/audiodevinfo.cc, libinterp/dldfcn/audioinfo.cc,
libinterp/dldfcn/audioread.cc, libinterp/dldfcn/audiowrite.cc,
libinterp/dldfcn/player_class.cc, libinterp/dldfcn/player_class.h,
libinterp/dldfcn/recorder_class.cc, libinterp/dldfcn/recorder_class.h,
scripts/audio/@audioplayer/__get_properties__.m,
scripts/audio/@audioplayer/audioplayer.m,
scripts/audio/@audioplayer/display.m, scripts/audio/@audioplayer/get.m,
scripts/audio/@audioplayer/isplaying.m, scripts/audio/@audioplayer/pause.m,
scripts/audio/@audioplayer/play.m,
scripts/audio/@audioplayer/playblocking.m,
scripts/audio/@audioplayer/resume.m, scripts/audio/@audioplayer/set.m,
scripts/audio/@audioplayer/stop.m, scripts/audio/@audioplayer/subsasgn.m,
scripts/audio/@audioplayer/subsref.m,
scripts/audio/@audiorecorder/__get_properties__.m,
scripts/audio/@audiorecorder/audiorecorder.m,
scripts/audio/@audiorecorder/display.m, scripts/audio/@audiorecorder/get.m,
scripts/audio/@audiorecorder/getaudiodata.m,
scripts/audio/@audiorecorder/getplayer.m,
scripts/audio/@audiorecorder/isrecording.m,
scripts/audio/@audiorecorder/pause.m, scripts/audio/@audiorecorder/play.m,
scripts/audio/@audiorecorder/record.m,
scripts/audio/@audiorecorder/recordblocking.m,
scripts/audio/@audiorecorder/resume.m, scripts/audio/@audiorecorder/set.m,
scripts/audio/@audiorecorder/stop.m,
scripts/audio/@audiorecorder/subsasgn.m,
scripts/audio/@audiorecorder/subsref.m: Apply consistent Octave indentation,
spacing, and quoting styles. Strip trailing whitespace. Remove braces from
one-line if-else blocks. Simplify some variable declarations.
author | Mike Miller <mtmiller@ieee.org> |
---|---|
date | Thu, 03 Oct 2013 07:52:58 -0400 |
parents | 1f551d169db2 |
children | ce02743b6f2a |
line wrap: on
line diff
--- a/libinterp/dldfcn/audioread.cc Wed Oct 02 00:14:09 2013 -0400 +++ b/libinterp/dldfcn/audioread.cc Thu Oct 03 07:52:58 2013 -0400 @@ -27,33 +27,33 @@ #include "oct.h" #include "ov-struct.h" #ifdef HAVE_SNDFILE - #include <sndfile.h> +#include <sndfile.h> #endif - -DEFUN_DLD(audioread, args, , + +DEFUN_DLD (audioread, args, , "-*- texinfo -*-\n\ -@deftypefn{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename})\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\ @end deftypefn\n\ -@deftypefn{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename}, @var{samples})\n\ +@deftypefn{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\ @end deftypefn\n\ -@deftypefn{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename}, @var{dataType})\n\ +@deftypefn{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\ @end deftypefn\n\ -@deftypefn{Loadable Function} [@var{y}, @var{Fs}] = audioread(@var{filename}, @var{samples}, @var{dataType})\n\ +@deftypefn{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\ @@ -68,11 +68,11 @@ SF_INFO info; info.format = 0; int start, end; - file = sf_open(args(0).string_value ().c_str (), SFM_READ, &info); + 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); + 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 (); @@ -86,59 +86,41 @@ { 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 (); - } + type = args(2).string_value (); else - { - type = args(1).string_value (); - } + type = args(1).string_value (); + if (type == "native") { if (info.format & SF_FORMAT_PCM_S8) - { - ret_audio = octave_value ((audio * 127)).int8_array_value (); - } + 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 (); - } + 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 (); - } + 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 (); - } + 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 (); - } + ret_audio = octave_value ((audio * 2147483648)).int32_array_value (); else - { - ret_audio = octave_value (audio); - } + ret_audio = octave_value (audio); } else - { - ret_audio = octave_value (audio); - } + ret_audio = octave_value (audio); } else - { - ret_audio = octave_value (audio); - } + ret_audio = octave_value (audio); + retval(0) = ret_audio; retval(1) = info.samplerate; #else - error("sndfile not found on your system and thus audioread is not functional"); + error ("sndfile not found on your system and thus audioread is not functional"); #endif - return octave_value(retval); + return octave_value (retval); }