265 lines
6.7 KiB
C
265 lines
6.7 KiB
C
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/*******************************************************************************/
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/* Copyright (C) 2008 Jonathan Moore Liles */
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/* */
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/* This program is free software; you can redistribute it and/or modify it */
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/* under the terms of the GNU General Public License as published by the */
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/* Free Software Foundation; either version 2 of the License, or (at your */
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/* option) any later version. */
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/* */
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/* This program is distributed in the hope that it will be useful, but WITHOUT */
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/* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or */
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/* FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for */
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/* more details. */
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/* */
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/* You should have received a copy of the GNU General Public License along */
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/* with This program; see the file COPYING. If not,write to the Free Software */
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/* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
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/*******************************************************************************/
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#include "Audio_File_SF.H"
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// #include "Timeline.H"
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#include <sndfile.h>
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#include <stdlib.h>
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#include <string.h>
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#include <assert.h>
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#include "Peaks.H"
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// #define HAS_SF_FORMAT_VORBIS
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#include "const.h"
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#include "debug.h"
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#include <stdio.h>
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const Audio_File::format_desc Audio_File_SF::supported_formats[] =
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{
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{ "Wav 24", "wav", SF_FORMAT_WAV | SF_FORMAT_PCM_24 | SF_ENDIAN_FILE },
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{ "Wav 16", "wav", SF_FORMAT_WAV | SF_FORMAT_PCM_16 | SF_ENDIAN_FILE },
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{ "Wav f32", "wav", SF_FORMAT_WAV | SF_FORMAT_FLOAT | SF_ENDIAN_FILE },
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{ "Au 24", "au", SF_FORMAT_AU | SF_FORMAT_PCM_24 | SF_ENDIAN_FILE },
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{ "Au 16", "au", SF_FORMAT_AU | SF_FORMAT_PCM_16 | SF_ENDIAN_FILE },
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{ "FLAC", "flac", SF_FORMAT_FLAC | SF_FORMAT_PCM_24 },
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#ifdef HAVE_SF_FORMAT_VORBIS
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{ "Vorbis q10", "ogg", SF_FORMAT_OGG | SF_FORMAT_VORBIS, 10 },
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{ "Vorbis q6", "ogg", SF_FORMAT_OGG | SF_FORMAT_VORBIS, 6 },
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{ "Vorbis q3", "ogg", SF_FORMAT_OGG | SF_FORMAT_VORBIS, 3 },
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#endif
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{ 0, 0 }
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};
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Audio_File_SF *
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Audio_File_SF::from_file ( const char *filename )
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{
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SNDFILE *in;
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SF_INFO si;
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Audio_File_SF *c = NULL;
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memset( &si, 0, sizeof( si ) );
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char *fp = path( filename );
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if ( ! ( in = sf_open( fp, SFM_READ, &si ) ) )
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return NULL;
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/* if ( si.samplerate != timeline->sample_rate() ) */
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/* { */
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/* printf( "error: samplerate mismatch!\n" ); */
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/* goto invalid; */
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/* } */
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c = new Audio_File_SF;
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// c->_peak_writer = NULL;
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c->_current_read = 0;
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c->_filename = strdup( filename );
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c->_path = fp;
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c->_length = si.frames;
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c->_samplerate = si.samplerate;
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c->_channels = si.channels;
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c->_in = in;
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// sf_close( in );
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return c;
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//invalid:
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sf_close( in );
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return NULL;
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}
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Audio_File_SF *
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Audio_File_SF::create ( const char *filename, nframes_t samplerate, int channels, const char *format )
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{
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SF_INFO si;
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SNDFILE *out;
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memset( &si, 0, sizeof( si ) );
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const Audio_File::format_desc *fd = Audio_File::find_format( Audio_File_SF::supported_formats, format );
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if ( ! fd )
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return (Audio_File_SF *)1;
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si.samplerate = samplerate;
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si.channels = channels;
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si.format = fd->id;
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char *name;
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asprintf( &name, "%s.%s", filename, fd->extension );
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char *filepath = path( name );
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if ( ! ( out = sf_open( filepath, SFM_WRITE, &si ) ) )
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{
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printf( "couldn't create soundfile.\n" );
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free( name );
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return NULL;
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}
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if ( !strcmp( fd->extension, "ogg" ) )
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{
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/* set high quality encoding for vorbis */
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double quality = ( fd->quality + 1 ) / (float)11;
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sf_command( out, SFC_SET_VBR_ENCODING_QUALITY, &quality, sizeof( double ) );
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}
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Audio_File_SF *c = new Audio_File_SF;
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c->_path = filepath;
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c->_filename = name;
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c->_length = 0;
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c->_samplerate = samplerate;
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c->_channels = channels;
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c->_in = out;
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c->_peaks.prepare_for_writing();
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return c;
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}
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bool
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Audio_File_SF::open ( void )
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{
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SF_INFO si;
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assert( _in == NULL );
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memset( &si, 0, sizeof( si ) );
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if ( ! ( _in = sf_open( _path, SFM_READ, &si ) ) )
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return false;
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_current_read = 0;
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_length = si.frames;
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_samplerate = si.samplerate;
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_channels = si.channels;
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// seek( 0 );
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return true;
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}
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void
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Audio_File_SF::close ( void )
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{
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if ( _in )
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sf_close( _in );
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_in = NULL;
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}
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void
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Audio_File_SF::seek ( nframes_t offset )
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{
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lock();
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if ( offset != _current_read )
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sf_seek( _in, _current_read = offset, SEEK_SET | SFM_READ );
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unlock();
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}
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/* if channels is -1, then all channels are read into buffer
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(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 )
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WARNING( "warning: attempt to read an insane number of frames (%lu) from soundfile\n", (unsigned long)len );
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// printf( "len = %lu, channels = %d\n", len, _channels );
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lock();
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nframes_t rlen;
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if ( _channels == 1 || channel == -1 )
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rlen = sf_readf_float( _in, buf, len );
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else
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{
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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 */
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for ( unsigned 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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}
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_current_read += rlen;
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unlock();
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return rlen;
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}
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/** read samples from /start/ to /end/ into /buf/ */
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nframes_t
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Audio_File_SF::read ( sample_t *buf, int channel, nframes_t start, nframes_t len )
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{
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lock();
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// open();
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seek( start );
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nframes_t cnt = read( buf, channel, len );
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unlock();
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// close();
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return cnt;
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}
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/** write /nframes/ from /buf/ to soundfile. Should be interleaved for
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* the appropriate number of channels */
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nframes_t
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Audio_File_SF::write ( sample_t *buf, nframes_t nframes )
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{
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_peaks.write( buf, nframes );
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lock();
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nframes_t l = sf_writef_float( _in, buf, nframes );
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_length += l;
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unlock();
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return l;
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}
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