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in draft-mode use zero-order-hold interpolation instead of linear interpolation - fixes some issues with resampling, cleanups

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@961 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2008-05-12 18:37:05 +00:00
parent 8cf16971b8
commit 298df38d09
5 changed files with 14 additions and 5 deletions
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9
ChangeLog
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@@ -1,5 +1,14 @@
2008-05-12 Tobias Doerffel <tobydox/at/users/dot/sourceforge/dot/net>
* plugins/bass_booster/bassbooster_controls.cpp:
* plugins/sf2_player/sf2_player.cpp:
* src/core/audio/audio_device.cpp:
cleanups and the like
* include/mixer.h:
in draft-mode use zero-order-hold interpolation instead of linear
interpolation - fixes some issues with resampling
* plugins/bass_booster/bassbooster_controls.cpp:
made bass-booster HQ-mode-capable

2
include/mixer.h
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@@ -177,7 +177,7 @@ public:
switch( interpolation )
{
case Interpolation_Linear:
return( SRC_LINEAR );
return( SRC_ZERO_ORDER_HOLD );
case Interpolation_SincFastest:
return( SRC_SINC_FASTEST );
case Interpolation_SincMedium:

3
plugins/bass_booster/bassbooster_controls.cpp
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@@ -57,7 +57,8 @@ bassBoosterControls::bassBoosterControls( bassBoosterEffect * _eff ) :
void bassBoosterControls::changeFrequency( void )
{
const sample_t fac = engine::getMixer()->processingSampleRate() / 44100;
const sample_t fac = engine::getMixer()->processingSampleRate() /
44100.0f;
m_effect->m_bbFX.leftFX().setFrequency( m_freqModel.value() * fac );
m_effect->m_bbFX.rightFX().setFrequency( m_freqModel.value() * fac );
}

2
plugins/sf2_player/sf2_player.cpp
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@@ -490,7 +490,7 @@ void sf2Instrument::play( bool _try_parallelizing,
src_data.data_out = _working_buffer[0];
src_data.input_frames = f;
src_data.output_frames = frames;
src_data.src_ratio = (float) frames / f;
src_data.src_ratio = (double) frames / f;
src_data.end_of_input = 0;
int error = src_process( m_srcState, &src_data );
delete[] tmp;

3
src/core/audio/audio_device.cpp
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@@ -168,18 +168,17 @@ void audioDevice::resample( const surroundSampleFrame * _src,
m_srcData.data_in = (float *) _src[0];
m_srcData.data_out = _dst[0];
m_srcData.src_ratio = (double) _dst_sr / _src_sr;
int error;
if( ( error = src_process( m_srcState, &m_srcData ) ) )
{
printf( "audioDevice::resample(): error while resampling: %s\n",
src_strerror( error ) );
}
m_srcData.end_of_input = 0;
}
Uint32 audioDevice::convertToS16( const surroundSampleFrame * _ab,
const fpp_t _frames,
const float _master_gain,