Mercurial > octave
view libinterp/dldfcn/audioread.cc @ 28223:45763d59cb4f stable
use nullptr instead of NULL or 0 in a few more places
* QWinTerminalImpl.cpp, oct-procbuf.cc, audioread.cc, jit-typeinfo.cc,
lo-sysdep.cc, url-transfer.cc, shared-fcns.h: Replace NULL and 0 with
nullptr where appropriate.
author | John W. Eaton <jwe@octave.org> |
---|---|
date | Wed, 15 Apr 2020 15:55:32 -0400 |
parents | bd51beb6205e |
children | 26cfccfee9a0 0a5b15007766 |
line wrap: on
line source
//////////////////////////////////////////////////////////////////////// // // Copyright (C) 2013-2020 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 #if defined (HAVE_SNDFILE) static void safe_close (SNDFILE *file) { sf_close (file); } #endif 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)); octave::unwind_protect frame; frame.add_fcn (safe_close, file); 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 = (octave::math::isinf (range(0)) ? info.frames : range(0)); double dend = (octave::math::isinf (range(1)) ? info.frames : range(1)); if (dstart < 1 || dstart > dend || dend > info.frames || octave::math::x_nint (dstart) != dstart || octave::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.fortran_vec (); 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; 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; 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; } 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 (octave::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)); octave::unwind_protect frame; frame.add_fcn (safe_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? 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"); octave::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)); octave::unwind_protect frame; frame.add_fcn (safe_close, file); octave_scalar_map result; std::string full_name = octave::sys::canonicalize_file_name (filename); result.assign ("Filename", full_name); result.assign ("CompressionMethod", ""); result.assign ("NumChannels", info.channels); result.assign ("SampleRate", info.samplerate); result.assign ("TotalSamples", info.frames); double dframes = info.frames; double drate = info.samplerate; result.assign ("Duration", dframes / drate); 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); } } #else octave_unused_parameter (args); err_disabled_feature ("audioformats", "getting sound formats through libsndfile"); #endif return octave_value_list (); }