non/Timeline/Audio_File_SF.C

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/*******************************************************************************/
/* Copyright (C) 2008 Jonathan Moore Liles */
/* */
/* This program 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 2 of the License, or (at your */
/* option) any later version. */
/* */
/* This program 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 This program; see the file COPYING. If not,write to the Free Software */
/* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
/*******************************************************************************/
#include "Audio_File_SF.H"
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// #include "Timeline.H"
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#include <sndfile.h>
#include <stdlib.h>
#include <string.h>
#include <assert.h>
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#include "Peaks.H"
// #define HAS_SF_FORMAT_VORBIS
const Audio_File::format_desc Audio_File_SF::supported_formats[] =
{
{ "Wav 24", "wav", SF_FORMAT_WAV | SF_FORMAT_PCM_24 | SF_ENDIAN_FILE },
{ "Wav 16", "wav", SF_FORMAT_WAV | SF_FORMAT_PCM_16 | SF_ENDIAN_FILE },
{ "Wav f32", "wav", SF_FORMAT_WAV | SF_FORMAT_FLOAT | SF_ENDIAN_FILE },
{ "Au 24", "au", SF_FORMAT_AU | SF_FORMAT_PCM_24 | SF_ENDIAN_FILE },
{ "Au 16", "au", SF_FORMAT_AU | SF_FORMAT_PCM_16 | SF_ENDIAN_FILE },
{ "FLAC", "flac", SF_FORMAT_FLAC | SF_FORMAT_PCM_24 },
#ifdef HAS_SF_FORMAT_VORBIS
{ "Ogg Vorbis", "ogg", SF_FORMAT_OGG | SF_FORMAT_VORBIS | SF_FORMAT_PCM_16 },
#endif
{ 0, 0 }
};
Audio_File_SF *
Audio_File_SF::from_file ( const char *filename )
{
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SNDFILE *in;
SF_INFO si;
Audio_File_SF *c = NULL;
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memset( &si, 0, sizeof( si ) );
if ( ! ( in = sf_open( filename, SFM_READ, &si ) ) )
{
printf( "couldn't open file\n" );
return NULL;
}
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/* if ( si.samplerate != timeline->sample_rate() ) */
/* { */
/* printf( "error: samplerate mismatch!\n" ); */
/* goto invalid; */
/* } */
c = new Audio_File_SF;
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c->_peak_writer = NULL;
c->_current_read = 0;
c->_filename = strdup( filename );
c->_length = si.frames;
c->_samplerate = si.samplerate;
c->_channels = si.channels;
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c->_in = in;
// sf_close( in );
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return c;
invalid:
sf_close( in );
return NULL;
}
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Audio_File_SF *
Audio_File_SF::create ( const char *filename, nframes_t samplerate, int channels, const char *format )
{
SF_INFO si;
SNDFILE *out;
memset( &si, 0, sizeof( si ) );
const Audio_File::format_desc *fd = Audio_File::find_format( Audio_File_SF::supported_formats, format );
if ( ! fd )
return (Audio_File_SF *)1;
si.samplerate = samplerate;
si.channels = channels;
si.format = fd->id;
char *name;
asprintf( &name, "%s.%s", filename, fd->extension );
// if ( ! ( out = sf_open( name, SFM_RDWR, &si ) ) )
if ( ! ( out = sf_open( name, SFM_WRITE, &si ) ) )
{
printf( "couldn't create soundfile.\n" );
free( name );
return NULL;
}
Audio_File_SF *c = new Audio_File_SF;
c->_filename = name;
c->_length = 0;
c->_samplerate = samplerate;
c->_channels = channels;
c->_in = out;
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/* FIXME: 256 ? */
c->_peak_writer = new Peak_Writer( name, 256, channels );
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return c;
}
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bool
Audio_File_SF::open ( void )
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{
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SF_INFO si;
assert( _in == NULL );
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memset( &si, 0, sizeof( si ) );
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if ( ! ( _in = sf_open( _filename, SFM_READ, &si ) ) )
return false;
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_current_read = 0;
_length = si.frames;
_samplerate = si.samplerate;
_channels = si.channels;
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// seek( 0 );
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return true;
}
void
Audio_File_SF::close ( void )
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{
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if ( _in )
sf_close( _in );
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if ( _peak_writer )
delete _peak_writer;
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_in = NULL;
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}
void
Audio_File_SF::seek ( nframes_t offset )
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{
if ( offset != _current_read )
{
sf_seek( _in, _current_read = offset, SEEK_SET | SFM_READ );
}
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}
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/* if channels is -1, then all channels are read into buffer
(interleaved). buf should be big enough to hold them all */
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nframes_t
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Audio_File_SF::read ( sample_t *buf, int channel, nframes_t len )
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{
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if ( len > 256 * 100 )
printf( "warning: attempt to read an insane number of frames (%lu) from soundfile\n", len );
// printf( "len = %lu, channels = %d\n", len, _channels );
nframes_t rlen;
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if ( _channels == 1 || channel == -1 )
rlen = sf_readf_float( _in, buf, len );
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else
{
sample_t *tmp = new sample_t[ len * _channels ];
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rlen = sf_readf_float( _in, tmp, len );
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/* extract the requested channel */
for ( int i = channel; i < rlen * _channels; i += _channels )
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*(buf++) = tmp[ i ];
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delete[] tmp;
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}
_current_read += rlen;
return rlen;
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}
/** read samples from /start/ to /end/ into /buf/ */
nframes_t
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Audio_File_SF::read ( sample_t *buf, int channel, nframes_t start, nframes_t end )
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{
assert( end > start );
// open();
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seek( start );
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nframes_t len = read( buf, channel, end - start );
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// close();
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return len;
}
/** write /nframes/ from /buf/ to soundfile. Should be interleaved for
* the appropriate number of channels */
nframes_t
Audio_File_SF::write ( sample_t *buf, nframes_t nframes )
{
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_peak_writer->write( buf, nframes );
nframes_t l = sf_writef_float( _in, buf, nframes );
_length += l;
return l;
}