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lot of bugfixes, especially in VST-support-layer and some nice new features

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@44 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2006-01-03 13:15:42 +00:00
parent f049e70334
commit 052224c8e2
72 changed files with 2103 additions and 871 deletions
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235
src/lib/sample_buffer.cpp
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@@ -1,7 +1,7 @@
/*
* sample_buffer.cpp - container-class sampleBuffer
*
* Copyright (c) 2005 Tobias Doerffel <tobydox/at/users.sourceforge.net>
* Copyright (c) 2005-2006 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
@@ -159,6 +159,35 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, Uint32 _frames ) :
sampleBuffer::sampleBuffer( Uint32 _frames ) :
QObject(),
m_audioFile( "" ),
m_origData( NULL ),
m_origFrames( 0 ),
m_data( NULL ),
m_frames( 0 ),
m_startFrame( 0 ),
m_endFrame( 0 ),
m_amplification( 1.0f ),
m_reversed( FALSE ),
m_dataMutex()
{
m_origData = new sampleFrame[_frames];
memset( m_origData, 0, _frames * BYTES_PER_FRAME );
m_origFrames = _frames;
#ifdef SDL_SDL_SOUND_H
// init sound-file-system of SDL
Sound_Init();
#endif
#ifdef HAVE_SAMPLERATE_H
initResampling();
#endif
update();
}
sampleBuffer::~sampleBuffer()
{
m_dataMutex.lock();
@@ -166,11 +195,12 @@ sampleBuffer::~sampleBuffer()
m_origData = NULL;
delete[] m_data;
m_data = NULL;
m_dataMutex.unlock();
#ifdef HAVE_SAMPLERATE_H
quitResampling();
#endif
m_dataMutex.unlock();
}
@@ -223,45 +253,36 @@ void sampleBuffer::update( bool _keep_settings )
file.ascii();
#endif
Sint16 * buf = NULL;
Uint8 channels;
Uint8 channels = DEFAULT_CHANNELS;
Uint32 samplerate = SAMPLE_RATES[DEFAULT_QUALITY_LEVEL];
#ifdef HAVE_SNDFILE_H
if( m_frames == 0 )
{
m_frames = decodeSampleSF( f, buf, channels );
}
#endif
#ifdef SDL_SDL_SOUND_H
if( m_frames == 0 )
{
m_frames = decodeSampleSDL( f, buf, channels );
m_frames = decodeSampleSF( f, buf, channels,
samplerate );
}
#endif
#ifdef HAVE_VORBIS_VORBISFILE_H
if( m_frames == 0 )
{
m_frames = decodeSampleOGG( f, buf, channels );
m_frames = decodeSampleOGGVorbis( f, buf, channels,
samplerate );
}
#endif
#ifdef SDL_SDL_SOUND_H
if( m_frames == 0 )
{
m_frames = decodeSampleSDL( f, buf, channels,
samplerate );
}
#endif
if( m_frames > 0 && buf != NULL )
{
if( _keep_settings == FALSE )
{
// update frame-variables
m_startFrame = 0;
if( m_frames > 0 )
{
m_endFrame = m_frames - 1;
}
else
{
m_endFrame = 0;
}
}
// following code transforms int-samples into
// float-samples and does amplifying & reversing
float fac = m_amplification / 32767.0f;
const float fac = m_amplification /
OUTPUT_SAMPLE_MULTIPLIER;
m_data = new sampleFrame[m_frames];
// if reversing is on, we also reverse when
@@ -297,9 +318,39 @@ m_data[frame][chnl] = buf[idx] * fac;
delete[] buf;
// do samplerate-conversion if sample-decoder didn't
// convert sample-rate to our default-samplerate
if( samplerate != SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
{
sampleBuffer * resampled = resample( this,
samplerate,
SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] );
delete[] m_data;
m_frames = resampled->frames();
m_data = new sampleFrame[m_frames];
memcpy( m_data, resampled->data(), m_frames *
sizeof( sampleFrame ) );
delete resampled;
}
if( _keep_settings == FALSE )
{
// update frame-variables
m_startFrame = 0;
if( m_frames > 0 )
{
m_endFrame = m_frames - 1;
}
else
{
m_endFrame = 0;
}
}
}
else
{
// sample couldn't be decoded, create buffer containing
// one sample-frame
m_data = new sampleFrame[1];
memset( m_data, 0, sizeof( *m_data ) );
m_frames = 1;
@@ -309,6 +360,8 @@ m_data[frame][chnl] = buf[idx] * fac;
}
else
{
// neither an audio-file nor a buffer to copy from, so create
// buffer containing one sample-frame
m_data = new sampleFrame[1];
memset( m_data, 0, sizeof( *m_data ) * 1 );
m_frames = 1;
@@ -326,13 +379,14 @@ m_data[frame][chnl] = buf[idx] * fac;
#ifdef SDL_SDL_SOUND_H
Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
Uint8 & _channels )
Uint8 & _channels,
Uint32 & _samplerate )
{
Sound_AudioInfo STD_AUDIO_INFO =
{
AUDIO_S16SYS,
DEFAULT_CHANNELS,
SAMPLE_RATES[DEFAULT_QUALITY_LEVEL]
_channels,
_samplerate,
} ;
Uint32 frames = 0;
@@ -343,7 +397,8 @@ Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
{
// let SDL_sound decode our file to requested format
( void )Sound_DecodeAll( snd_sample );
_channels = STD_AUDIO_INFO.channels;
_channels = snd_sample->actual.channels;
_samplerate = snd_sample->actual.rate;
frames = snd_sample->buffer_size / ( BYTES_PER_OUTPUT_SAMPLE *
_channels );
_buf = new Sint16[frames * _channels];
@@ -360,7 +415,8 @@ Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
#ifdef HAVE_SNDFILE_H
Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
Uint8 & _channels )
Uint8 & _channels,
Uint32 & _samplerate )
{
SNDFILE * snd_file;
SF_INFO sf_info;
@@ -377,6 +433,7 @@ Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
_buf = new Sint16[sf_info.channels * frames];
frames = sf_read_short( snd_file, _buf, frames );
_channels = sf_info.channels;
_samplerate = sf_info.samplerate;
sf_close( snd_file );
}
@@ -384,7 +441,7 @@ Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
{
#ifdef DEBUG_LMMS
printf( "sampleBuffer::decodeSampleSF(): could not load "
"sample %s: %s\n", _f, sf_strerror( NULL ) );
"sample %s: %s\n", _f, sf_strerror( NULL ) );
#endif
}
return( frames );
@@ -398,9 +455,6 @@ Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
// callback-functions for reading ogg-file
#ifndef QT4
#endif
size_t qfileReadCallback( void * _ptr, size_t _size, size_t _n, void * _udata )
{
return( static_cast<QFile *>( _udata )->read( (char*)_ptr,
@@ -449,8 +503,9 @@ long qfileTellCallback( void * _udata )
Uint32 sampleBuffer::decodeSampleOGG( const char * _f, Sint16 * & _buf,
Uint8 & _channels )
Uint32 sampleBuffer::decodeSampleOGGVorbis( const char * _f, Sint16 * & _buf,
Uint8 & _channels,
Uint32 & _samplerate )
{
static ov_callbacks callbacks =
{
@@ -482,20 +537,20 @@ Uint32 sampleBuffer::decodeSampleOGG( const char * _f, Sint16 * & _buf,
switch( err )
{
case OV_EREAD:
printf( "sampleBuffer::decodeSampleOgg(): "
"media read error\n" );
printf( "sampleBuffer::decodeSampleOGGVorbis():"
" media read error\n" );
break;
case OV_ENOTVORBIS:
printf( "sampleBuffer::decodeSampleOgg(): "
"not an Ogg Vorbis file\n" );
/* printf( "sampleBuffer::decodeSampleOGGVorbis():"
" not an Ogg Vorbis file\n" );*/
break;
case OV_EVERSION:
printf( "sampleBuffer::decodeSampleOgg(): "
"vorbis version mismatch\n" );
printf( "sampleBuffer::decodeSampleOGGVorbis():"
" vorbis version mismatch\n" );
break;
case OV_EBADHEADER:
printf( "sampleBuffer::decodeSampleOgg(): "
"invalid Vorbis bitstream header\n" );
printf( "sampleBuffer::decodeSampleOGGVorbis():"
" invalid Vorbis bitstream header\n" );
break;
case OV_EFAULT:
printf( "sampleBuffer::decodeSampleOgg(): "
@@ -509,6 +564,8 @@ Uint32 sampleBuffer::decodeSampleOGG( const char * _f, Sint16 * & _buf,
ov_pcm_seek( &vf, 0 );
_channels = ov_info( &vf, -1 )->channels;
_samplerate = ov_info( &vf, -1 )->rate;
ogg_int64_t total = ov_pcm_total( &vf, -1 );
_buf = new Sint16[total * _channels];
@@ -517,7 +574,7 @@ Uint32 sampleBuffer::decodeSampleOGG( const char * _f, Sint16 * & _buf,
do
{
bytes_read = ov_read( &vf, (char *)&_buf[frames * _channels],
bytes_read = ov_read( &vf, (char *) &_buf[frames * _channels],
( total - frames ) * _channels *
sizeof( Sint16 ),
isLittleEndian()? 0 : 1,
@@ -694,7 +751,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
total_frames_for_current_pitch;
}
const float src_frame_idx = frame*freq_factor;
const float src_frame_idx = frame * freq_factor;
Uint32 frame_num = static_cast<Uint32>(
src_frame_idx) - src_frame_base;
const float frac_pos = src_frame_idx -
@@ -912,35 +969,35 @@ QString sampleBuffer::openAudioFile( void ) const
// set filters
#ifdef QT4
QStringList types;
types << tr( "All Audio-Files (*.wav *.ogg *.voc *.aif *.aiff *.au "
"*.raw)" )
types << tr( "All Audio-Files (*.wav *.ogg *.flac *.voc *.aif *.aiff "
"*.au *.raw)" )
<< tr( "Wave-Files (*.wav)" )
<< tr( "OGG-Files (*.ogg)" )
<< tr( "FLAC-Files (*.flac)" )
//<< tr( "MP3-Files (*.mp3)" )
//<< tr( "MIDI-Files (*.mid)" )
<< tr( "VOC-Files (*.voc)" )
<< tr( "AIFF-Files (*.aif *.aiff)" )
<< tr( "AU-Files (*.au)" )
<< tr( "RAW-Files (*.raw)" )
//<< tr( "FLAC-Files (*.flac)" )
//<< tr( "MOD-Files (*.mod)" )
;
ofd.setFilters( types );
#else
ofd.addFilter( tr( "All Audio-Files (*.wav *.ogg *.voc *.aif *.aiff "
"*.au *.raw)" ) );
ofd.addFilter( tr( "All Audio-Files (*.wav *.ogg *.flac *.voc *.aif "
"*.aiff *.au *.raw)" ) );
ofd.addFilter( tr( "Wave-Files (*.wav)" ) );
ofd.addFilter( tr( "OGG-Files (*.ogg)" ) );
ofd.addFilter( tr( "FLAC-Files (*.flac)" ) );
//ofd.addFilter (tr("MP3-Files (*.mp3)"));
//ofd.addFilter (tr("MIDI-Files (*.mid)"));
//ofd.addFilter (tr("MIDI-Files (*.mid)"));^
ofd.addFilter( tr( "VOC-Files (*.voc)" ) );
ofd.addFilter( tr( "AIFF-Files (*.aif *.aiff)" ) );
ofd.addFilter( tr( "AU-Files (*.au)" ) );
ofd.addFilter( tr( "RAW-Files (*.raw)" ) );
//ofd.addFilter (tr("FLAC-Files (*.flac)"));
//ofd.addFilter (tr("MOD-Files (*.mod)"));
ofd.setSelectedFilter( tr( "All Audio-Files (*.wav *.ogg *.voc *.aif "
"*.aiff *.au *.raw)" ) );
ofd.setSelectedFilter( tr( "All Audio-Files (*.wav *.ogg *.flac *.voc "
"*.aif *.aiff *.au *.raw)" ) );
#endif
if( m_audioFile != "" )
{
@@ -1016,6 +1073,8 @@ QString sampleBuffer::toBase64( void ) const
{
return( "" );
}
// the following code is quite confusing as we have to handle four
// different configs (no flac/qt3, no flac/qt4, flac/qt3, flac/qt4)
#ifdef HAVE_FLAC_STREAM_ENCODER_H
const Uint32 FRAMES_PER_BUF = 1152;
@@ -1155,6 +1214,69 @@ QString sampleBuffer::toBase64( void ) const
sampleBuffer * sampleBuffer::resample( sampleFrame * _data,
const Uint32 _frames,
const Uint32 _src_sr,
const Uint32 _dst_sr )
{
const Uint32 dst_frames = static_cast<Uint32>( _frames /
(float) _src_sr * (float) _dst_sr );
sampleBuffer * dst_sb = new sampleBuffer( dst_frames );
sampleFrame * dst_buf = dst_sb->m_origData;
#ifdef HAVE_SAMPLERATE_H
// yeah, libsamplerate, let's rock with sinc-interpolation!
int error;
SRC_STATE * state;
if( ( state = src_new( SRC_SINC_MEDIUM_QUALITY,
DEFAULT_CHANNELS, &error ) ) != NULL )
{
SRC_DATA src_data;
src_data.end_of_input = 0;
src_data.data_in = _data[0];
src_data.data_out = dst_buf[0];
src_data.input_frames = _frames;
src_data.output_frames = dst_frames;
src_data.src_ratio = (float) _dst_sr / _src_sr;
int error;
if( ( error = src_process( state, &src_data ) ) )
{
printf( "sampleBuffer: error while resampling: %s\n",
src_strerror( error ) );
}
src_delete( state );
}
else
{
printf( "Error: src_new() failed in sample_buffer.cpp!\n" );
}
#else
// no libsamplerate, so do simple cubic interpolation
for( Uint32 frame = 0; frame < dst_frames; ++frame )
{
const float src_frame_float = frame * (float) _src_sr / _dst_sr;
const float frac_pos = src_frame_float -
static_cast<Uint32>( src_frame_float );
const Uint32 src_frame = tLimit<Uint32>(
static_cast<Uint32>( src_frame_float ),
1, _frames - 2 );
for( Uint8 ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
dst_buf[frame][ch] = cubicInterpolate(
_data[src_frame - 1][ch],
_data[src_frame + 0][ch],
_data[src_frame + 1][ch],
_data[src_frame + 2][ch],
frac_pos );
}
}
#endif
dst_sb->update();
return( dst_sb );
}
void sampleBuffer::setAudioFile( const QString & _audio_file )
{
m_audioFile = _audio_file;
@@ -1375,6 +1497,7 @@ void sampleBuffer::loadFromBase64( const QString & _data )
delete[] m_origData;
m_origData = new sampleFrame[m_origFrames];
memcpy( m_origData, orig_data.data(), orig_data.size() );
m_audioFile = "";
update();
#ifndef QT4
// delete[] dst;