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better VST support and bugfixes

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@77 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2006-02-06 12:52:58 +00:00
parent af3d57b67f
commit de00678815
43 changed files with 966 additions and 668 deletions
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170
src/lib/sample_buffer.cpp
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@@ -131,7 +131,7 @@ sampleBuffer::sampleBuffer( const QString & _audio_file,
sampleBuffer::sampleBuffer( const sampleFrame * _data, Uint32 _frames ) :
sampleBuffer::sampleBuffer( const sampleFrame * _data, const f_cnt_t _frames ) :
QObject(),
m_audioFile( "" ),
m_origData( NULL ),
@@ -160,7 +160,7 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, Uint32 _frames ) :
sampleBuffer::sampleBuffer( Uint32 _frames ) :
sampleBuffer::sampleBuffer( const f_cnt_t _frames ) :
QObject(),
m_audioFile( "" ),
m_origData( NULL ),
@@ -253,9 +253,9 @@ void sampleBuffer::update( bool _keep_settings )
#else
file.ascii();
#endif
Sint16 * buf = NULL;
Uint8 channels = DEFAULT_CHANNELS;
Uint32 samplerate = SAMPLE_RATES[DEFAULT_QUALITY_LEVEL];
int_sample_t * buf = NULL;
ch_cnt_t channels = DEFAULT_CHANNELS;
sample_rate_t samplerate = SAMPLE_RATES[DEFAULT_QUALITY_LEVEL];
#ifdef HAVE_SNDFILE_H
if( m_frames == 0 )
@@ -290,28 +290,28 @@ void sampleBuffer::update( bool _keep_settings )
// scaling
if( m_reversed )
{
for( Uint32 frame = 0; frame < m_frames;
for( f_cnt_t frame = 0; frame < m_frames;
++frame )
{
for( Uint8 chnl = 0;
for( ch_cnt_t chnl = 0;
chnl < DEFAULT_CHANNELS;
++chnl )
{
const Uint32 idx = ( m_frames - frame ) * channels + ( chnl % channels );
const f_cnt_t idx = ( m_frames - frame ) * channels + ( chnl % channels );
m_data[frame][chnl] = buf[idx] * fac;
}
}
}
else
{
for( Uint32 frame = 0; frame < m_frames;
for( f_cnt_t frame = 0; frame < m_frames;
++frame )
{
for( Uint8 chnl = 0;
for( ch_cnt_t chnl = 0;
chnl < DEFAULT_CHANNELS;
++chnl )
{
const Uint32 idx = frame * channels + ( chnl % channels );
const f_cnt_t idx = frame * channels + ( chnl % channels );
m_data[frame][chnl] = buf[idx] * fac;
}
}
@@ -379,9 +379,10 @@ m_data[frame][chnl] = buf[idx] * fac;
#ifdef SDL_SDL_SOUND_H
Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
Uint8 & _channels,
Uint32 & _samplerate )
f_cnt_t sampleBuffer::decodeSampleSDL( const char * _f,
int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _samplerate )
{
Sound_AudioInfo STD_AUDIO_INFO =
{
@@ -389,7 +390,7 @@ Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
_channels,
_samplerate,
} ;
Uint32 frames = 0;
f_cnt_t frames = 0;
Sound_Sample * snd_sample = Sound_NewSampleFromFile( _f,
&STD_AUDIO_INFO, 16384 );
@@ -400,9 +401,9 @@ Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
( void )Sound_DecodeAll( snd_sample );
_channels = snd_sample->actual.channels;
_samplerate = snd_sample->actual.rate;
frames = snd_sample->buffer_size / ( BYTES_PER_OUTPUT_SAMPLE *
frames = snd_sample->buffer_size / ( BYTES_PER_INT_SAMPLE *
_channels );
_buf = new Sint16[frames * _channels];
_buf = new int_sample_t[frames * _channels];
memcpy( _buf, snd_sample->buffer, snd_sample->buffer_size );
Sound_FreeSample( snd_sample );
@@ -415,13 +416,14 @@ Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
#ifdef HAVE_SNDFILE_H
Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
Uint8 & _channels,
Uint32 & _samplerate )
f_cnt_t sampleBuffer::decodeSampleSF( const char * _f,
int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _samplerate )
{
SNDFILE * snd_file;
SF_INFO sf_info;
Uint32 frames = 0;
f_cnt_t frames = 0;
#ifdef OLD_SNDFILE
if( ( snd_file = sf_open_read( _f, &sf_info ) ) != NULL )
{
@@ -431,7 +433,7 @@ Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
{
frames = sf_info.frames;
#endif
_buf = new Sint16[sf_info.channels * frames];
_buf = new int_sample_t[sf_info.channels * frames];
frames = sf_read_short( snd_file, _buf, frames );
_channels = sf_info.channels;
_samplerate = sf_info.samplerate;
@@ -458,7 +460,7 @@ Uint32 sampleBuffer::decodeSampleSF( const char * _f, Sint16 * & _buf,
size_t qfileReadCallback( void * _ptr, size_t _size, size_t _n, void * _udata )
{
return( static_cast<QFile *>( _udata )->read( (char*)_ptr,
return( static_cast<QFile *>( _udata )->read( (char*) _ptr,
_size * _n ) );
}
@@ -504,9 +506,10 @@ long qfileTellCallback( void * _udata )
Uint32 sampleBuffer::decodeSampleOGGVorbis( const char * _f, Sint16 * & _buf,
Uint8 & _channels,
Uint32 & _samplerate )
f_cnt_t sampleBuffer::decodeSampleOGGVorbis( const char * _f,
int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _samplerate )
{
static ov_callbacks callbacks =
{
@@ -518,7 +521,7 @@ Uint32 sampleBuffer::decodeSampleOGGVorbis( const char * _f, Sint16 * & _buf,
OggVorbis_File vf;
Uint32 frames = 0;
f_cnt_t frames = 0;
QFile * f = new QFile( _f );
#ifdef QT4
@@ -569,7 +572,7 @@ Uint32 sampleBuffer::decodeSampleOGGVorbis( const char * _f, Sint16 * & _buf,
ogg_int64_t total = ov_pcm_total( &vf, -1 );
_buf = new Sint16[total * _channels];
_buf = new int_sample_t[total * _channels];
int bitstream = 0;
long bytes_read = 0;
@@ -577,14 +580,14 @@ Uint32 sampleBuffer::decodeSampleOGGVorbis( const char * _f, Sint16 * & _buf,
{
bytes_read = ov_read( &vf, (char *) &_buf[frames * _channels],
( total - frames ) * _channels *
sizeof( Sint16 ),
isLittleEndian()? 0 : 1,
sizeof( Sint16 ), 1, &bitstream );
BYTES_PER_INT_SAMPLE,
isLittleEndian() ? 0 : 1,
BYTES_PER_INT_SAMPLE, 1, &bitstream );
if( bytes_read < 0 )
{
break;
}
frames += bytes_read / ( _channels * sizeof( Sint16 ) );
frames += bytes_read / ( _channels * BYTES_PER_INT_SAMPLE );
}
while( bytes_read != 0 && bitstream == 0 );
@@ -642,10 +645,12 @@ void sampleBuffer::destroyResamplingContext( SRC_STATE * _context )
bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
Uint32 _frames, float _freq,
bool _looped,
void * * _resampling_data )
bool FASTCALL sampleBuffer::play( sampleFrame * _ab,
const f_cnt_t _start_frame,
const fpab_t _frames,
const float _freq,
const bool _looped,
void * * _resampling_data )
{
mixer::inst()->clearAudioBuffer( _ab, _frames );
@@ -657,10 +662,10 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
const double freq_factor = (double) _freq / (double) BASE_FREQ;
const Sint16 freq_diff = static_cast<Sint16>( BASE_FREQ - _freq );
Uint32 frames_to_process = _frames;
fpab_t frames_to_process = _frames;
// calculate how many frames we have in requested pitch
const Uint32 total_frames_for_current_pitch = static_cast<Uint32>( (
const f_cnt_t total_frames_for_current_pitch = static_cast<f_cnt_t>( (
m_endFrame - m_startFrame ) /
freq_factor );
if( total_frames_for_current_pitch == 0 )
@@ -676,11 +681,11 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
}
// this holds the number of the first frame to play
const Uint32 play_frame = m_startFrame + ( _start_frame %
const f_cnt_t play_frame = m_startFrame + ( _start_frame %
total_frames_for_current_pitch );
// this holds the number of remaining frames in current loop
Uint32 frames_for_loop = total_frames_for_current_pitch -
f_cnt_t frames_for_loop = total_frames_for_current_pitch -
( play_frame - m_startFrame );
// make sure, data isn't accessed in any other way (e.g. deleting
@@ -691,7 +696,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
{
frames_to_process = frames_for_loop;
}
const Uint32 f1 = static_cast<Uint32>( play_frame * freq_factor );
const f_cnt_t f1 = static_cast<f_cnt_t>( play_frame * freq_factor );
/* Uint32 f2 = 0;
while( f2 < f1 )
{
@@ -723,7 +728,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
}
m_srcData.data_in = start_frame[0];
m_srcData.data_out = _ab[0];
m_srcData.input_frames = static_cast<Uint32>( frames_for_loop *
m_srcData.input_frames = static_cast<f_cnt_t>( frames_for_loop *
freq_factor );
m_srcData.output_frames = frames_to_process;
m_srcData.src_ratio = 1.0 / freq_factor;
@@ -734,12 +739,12 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
src_strerror( error ) );
}
#else
Uint32 src_frame_base = 0;
f_cnt_t src_frame_base = 0;
// check whether we're in high-quality-mode
if( mixer::inst()->highQuality() == TRUE )
{
// we are, so let's use cubic interpolation...
for( Uint32 frame = 0; frame < frames_to_process;
for( f_cnt_t frame = 0; frame < frames_to_process;
++frame )
{
// current loop done?
@@ -753,10 +758,10 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
}
const float src_frame_idx = frame * freq_factor;
Uint32 frame_num = static_cast<Uint32>(
f_cnt_t frame_num = static_cast<f_cnt_t>(
src_frame_idx) - src_frame_base;
const float frac_pos = src_frame_idx -
static_cast<Uint32>( src_frame_idx );
static_cast<f_cnt_t>( src_frame_idx );
// because of cubic interpolation we have to
// access start_frame[frame_num-1], so make
@@ -766,7 +771,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
{
frame_num = 1;
}
for ( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS;
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS;
++chnl )
{
_ab[frame][chnl] = cubicInterpolate(
@@ -782,10 +787,10 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
{
// just normal mode, so we can use linear
// interpolation...
for( Uint32 frame = 0; frame < frames_to_process;
for( f_cnt_t frame = 0; frame < frames_to_process;
++frame )
{
if( _looped && ( frame-src_frame_base ) >
if( _looped && ( frame - src_frame_base ) >
frames_for_loop )
{
start_frame = loop_start;
@@ -794,11 +799,11 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
total_frames_for_current_pitch;
}
const float src_frame_idx = frame * freq_factor;
const Uint32 frame_num = (Uint32)src_frame_idx -
src_frame_base + 0;
const f_cnt_t frame_num =
(f_cnt_t)src_frame_idx-src_frame_base;
const float frac_pos = src_frame_idx -
(Uint32) src_frame_idx;
for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS;
(f_cnt_t) src_frame_idx;
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS;
++chnl )
{
_ab[frame][chnl] = linearInterpolate(
@@ -816,7 +821,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
// as is into pitched-copy-buffer
if( _looped && frames_for_loop < frames_to_process )
{
Uint32 total_frames_copied = 0;
f_cnt_t total_frames_copied = 0;
while( total_frames_copied < frames_to_process )
{
memcpy( _ab[total_frames_copied], start_frame,
@@ -878,13 +883,14 @@ void sampleBuffer::drawWaves( QPainter & _p, QRect _dr, drawMethods _dm )
float old_x = _dr.x();
float old_y[DEFAULT_CHANNELS] = { y_base, y_base };
float fpp = tMax<float>( tMin<float>( m_frames / (float)w,
const float fpp = tMax<float>( tMin<float>( m_frames / (float)w,
20.0f ), 1.0f );
for( float frame = 0; frame < m_frames; frame += fpp )
{
const float x = frame*w / m_frames + _dr.x();
for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS;
++chnl )
{
const float y = y_base +
m_data[static_cast<int>( frame )][chnl] * y_space;
@@ -898,15 +904,15 @@ void sampleBuffer::drawWaves( QPainter & _p, QRect _dr, drawMethods _dm )
int old_x = _dr.x();
int old_y[DEFAULT_CHANNELS] = { y_base, y_base };
Uint32 fpp = tMax<Uint32>( tMin<Uint32>( m_frames / w, 20 ),
1 );
const f_cnt_t fpp = tMax<f_cnt_t>( tMin<f_cnt_t>( m_frames / w,
20 ), 1 );
for( Uint32 frame = 0; frame < m_frames; frame += fpp )
for( f_cnt_t frame = 0; frame < m_frames; frame += fpp )
{
const int x = static_cast<int>( frame /
(float) m_frames * w ) +
_dr.x();
for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
{
const Uint16 y = y_base +
static_cast<Uint16>( m_data[frame][chnl] * y_space );
@@ -920,10 +926,10 @@ void sampleBuffer::drawWaves( QPainter & _p, QRect _dr, drawMethods _dm )
}
else if( _dm == DOTS )
{
for( Uint32 frame = 0; frame < m_frames; ++frame )
for( f_cnt_t frame = 0; frame < m_frames; ++frame )
{
const int x = frame*w / m_frames + _dr.x();
for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
const int x = frame * w / m_frames + _dr.x();
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
{
_p.drawPoint( x, y_base +
static_cast<Uint16>( m_data[frame][chnl] * y_space ) );
@@ -1077,7 +1083,7 @@ QString & sampleBuffer::toBase64( QString & _dst ) const
}
#ifdef HAVE_FLAC_STREAM_ENCODER_H
const Uint32 FRAMES_PER_BUF = 1152;
const f_cnt_t FRAMES_PER_BUF = 1152;
FLAC__StreamEncoder * flac_enc = FLAC__stream_encoder_new();
FLAC__stream_encoder_set_channels( flac_enc, DEFAULT_CHANNELS );
@@ -1102,15 +1108,15 @@ QString & sampleBuffer::toBase64( QString & _dst ) const
{
printf( "error within FLAC__stream_encoder_init()!\n" );
}
Uint32 frame_cnt = 0;
f_cnt_t frame_cnt = 0;
while( frame_cnt < m_frames )
{
Uint32 remaining = tMin<Uint32>( FRAMES_PER_BUF,
f_cnt_t remaining = tMin<f_cnt_t>( FRAMES_PER_BUF,
m_frames - frame_cnt );
FLAC__int32 buf[FRAMES_PER_BUF * DEFAULT_CHANNELS];
for( Uint32 f = 0; f < remaining; ++f )
for( f_cnt_t f = 0; f < remaining; ++f )
{
for( Uint8 ch = 0; ch < DEFAULT_CHANNELS; ++ch )
for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
buf[f*DEFAULT_CHANNELS+ch] = (FLAC__int32)(
mixer::clip( m_data[f+frame_cnt][ch] ) *
@@ -1144,11 +1150,11 @@ QString & sampleBuffer::toBase64( QString & _dst ) const
sampleBuffer * sampleBuffer::resample( sampleFrame * _data,
const Uint32 _frames,
const Uint32 _src_sr,
const Uint32 _dst_sr )
const f_cnt_t _frames,
const sample_rate_t _src_sr,
const sample_rate_t _dst_sr )
{
const Uint32 dst_frames = static_cast<Uint32>( _frames /
const f_cnt_t dst_frames = static_cast<f_cnt_t>( _frames /
(float) _src_sr * (float) _dst_sr );
sampleBuffer * dst_sb = new sampleBuffer( dst_frames );
sampleFrame * dst_buf = dst_sb->m_origData;
@@ -1165,7 +1171,7 @@ sampleBuffer * sampleBuffer::resample( sampleFrame * _data,
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;
src_data.src_ratio = (double) _dst_sr / _src_sr;
int error;
if( ( error = src_process( state, &src_data ) ) )
{
@@ -1180,15 +1186,15 @@ sampleBuffer * sampleBuffer::resample( sampleFrame * _data,
}
#else
// no libsamplerate, so do simple cubic interpolation
for( Uint32 frame = 0; frame < dst_frames; ++frame )
for( f_cnt_t 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 ),
static_cast<f_cnt_t>( src_frame_float );
const f_cnt_t src_frame = tLimit<f_cnt_t>(
static_cast<f_cnt_t>( src_frame_float ),
1, _frames - 2 );
for( Uint8 ch = 0; ch < DEFAULT_CHANNELS; ++ch )
for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
dst_buf[frame][ch] = cubicInterpolate(
_data[src_frame - 1][ch],
@@ -1282,8 +1288,8 @@ FLAC__StreamDecoderWriteStatus flacStreamDecoderWriteCallback(
return( FLAC__STREAM_DECODER_WRITE_STATUS_ABORT );
}
const Uint32 frames = _frame->header.blocksize;
for( Uint32 frame = 0; frame < frames; ++frame )
const f_cnt_t frames = _frame->header.blocksize;
for( f_cnt_t frame = 0; frame < frames; ++frame )
{
sampleFrame sframe = { _buffer[0][frame] /
OUTPUT_SAMPLE_MULTIPLIER,
@@ -1397,7 +1403,7 @@ void sampleBuffer::loadFromBase64( const QString & _data )
void sampleBuffer::setStartFrame( Uint32 _s )
void sampleBuffer::setStartFrame( const f_cnt_t _s )
{
// don't set this parameter while playing
m_dataMutex.lock();
@@ -1408,7 +1414,7 @@ void sampleBuffer::setStartFrame( Uint32 _s )
void sampleBuffer::setEndFrame( Uint32 _e )
void sampleBuffer::setEndFrame( const f_cnt_t _e )
{
// don't set this parameter while playing
m_dataMutex.lock();