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removed already broken support for building LMMS without libsamplerate

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@499 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2007-07-29 20:33:13 +00:00
parent eb99ab60db
commit 22f1720701
7 changed files with 32 additions and 316 deletions
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126
src/lib/sample_buffer.cpp
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@@ -120,9 +120,7 @@ sampleBuffer::sampleBuffer( const QString & _audio_file,
// init sound-file-system of SDL
Sound_Init();
#endif
#ifdef HAVE_SAMPLERATE_H
initResampling();
#endif
if( _is_base64_data == TRUE )
{
loadFromBase64( _audio_file );
@@ -156,9 +154,7 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, const f_cnt_t _frames ) :
// init sound-file-system of SDL
Sound_Init();
#endif
#ifdef HAVE_SAMPLERATE_H
initResampling();
#endif
update();
}
@@ -188,9 +184,7 @@ sampleBuffer::sampleBuffer( const f_cnt_t _frames ) :
// init sound-file-system of SDL
Sound_Init();
#endif
#ifdef HAVE_SAMPLERATE_H
initResampling();
#endif
update();
}
@@ -591,12 +585,10 @@ f_cnt_t sampleBuffer::decodeSampleOGGVorbis( const char * _f,
#ifdef HAVE_SAMPLERATE_H
void sampleBuffer::initResampling( void )
{
m_srcData.end_of_input = 0;
}
#endif
@@ -656,24 +648,9 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, handleState * _state,
}
}
/* Uint32 f2 = 0;
while( f2 < f1 )
{
f2 += frames_to_process * freq_factor;
}
if( f2 > f1 && f2 >= frames_to_process )
{
f2 -= frames_to_process * freq_factor;
}*/
// static int foo = 0;
// calc pointer of first frame
//printf("diff:%d %f %d f2: %d input: %d\n", f2 -foo, play_frame * freq_factor, static_cast<Uint32>( play_frame * freq_factor ), f2, (Uint32)( frames_for_loop * freq_factor ) );
// foo = f2;
// check whether we have to change pitch...
if( freq_factor != 1.0 || _state->m_varying_pitch )
{
#ifdef HAVE_SAMPLERATE_H
// Generate output
const f_cnt_t margin = 64;
f_cnt_t fragment_size = (f_cnt_t)( _frames * freq_factor )
@@ -702,82 +679,6 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, handleState * _state,
{
play_frame = getLoopedIndex( play_frame );
}
#else
f_cnt_t src_frame_base = 0;
// check whether we're in high-quality-mode
if( engine::getMixer()->highQuality() == TRUE )
{
// we are, so let's use cubic interpolation...
for( f_cnt_t frame = 0; frame < frames_to_process;
++frame )
{
// current loop done?
if( _looped && ( frame-src_frame_base ) >
frames_for_loop )
{
start_frame = loop_start;
src_frame_base = frame;
frames_for_loop = frames_to_process %
total_frames_for_current_pitch;
}
const float src_frame_idx = frame * freq_factor;
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<f_cnt_t>( src_frame_idx );
// because of cubic interpolation we have to
// access start_frame[frame_num-1], so make
// sure we don't access data out of
// buffer-array-boundaries
if( frame_num == 0 && play_frame == 0 )
{
frame_num = 1;
}
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS;
++chnl )
{
_ab[frame][chnl] = cubicInterpolate(
start_frame[frame_num-1][chnl],
start_frame[frame_num+0][chnl],
start_frame[frame_num+1][chnl],
start_frame[frame_num+2][chnl],
frac_pos );
}
}
}
else
{
// just normal mode, so we can use linear
// interpolation...
for( f_cnt_t frame = 0; frame < frames_to_process;
++frame )
{
if( _looped && ( frame - src_frame_base ) >
frames_for_loop )
{
start_frame = loop_start;
src_frame_base = frame;
frames_for_loop = frames_to_process %
total_frames_for_current_pitch;
}
const float src_frame_idx = frame * freq_factor;
const f_cnt_t frame_num =
(f_cnt_t)src_frame_idx-src_frame_base;
const float frac_pos = src_frame_idx -
(f_cnt_t) src_frame_idx;
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS;
++chnl )
{
_ab[frame][chnl] = linearInterpolate(
start_frame[frame_num][chnl],
start_frame[frame_num+1][chnl],
frac_pos );
}
}
}
#endif
}
else
{
@@ -1162,7 +1063,7 @@ sampleBuffer * sampleBuffer::resample( sampleFrame * _data,
(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;
@@ -1188,27 +1089,6 @@ sampleBuffer * sampleBuffer::resample( sampleFrame * _data,
{
printf( "Error: src_new() failed in sample_buffer.cpp!\n" );
}
#else
// no libsamplerate, so do simple cubic interpolation
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<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 - 3 );
for( ch_cnt_t 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 );
}
@@ -1476,7 +1356,6 @@ sampleBuffer::handleState::handleState( bool _varying_pitch ) :
m_frame_index( 0 ),
m_varying_pitch( _varying_pitch )
{
#ifdef HAVE_SAMPLERATE_H
int error;
if( ( m_resampling_data = src_new(/*
( engine::getMixer()->highQuality() == TRUE ) ?
@@ -1486,7 +1365,6 @@ sampleBuffer::handleState::handleState( bool _varying_pitch ) :
{
printf( "Error: src_new() failed in sample_buffer.cpp!\n" );
}
#endif
}
@@ -1494,9 +1372,7 @@ sampleBuffer::handleState::handleState( bool _varying_pitch ) :
sampleBuffer::handleState::~handleState()
{
#ifdef HAVE_SAMPLERATE_H
src_delete( m_resampling_data );
#endif
}