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use new resampling-methods in case we're processing plugins which are known not to work at higher samplerates

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@970 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2008-05-13 17:45:09 +00:00
parent b2fc03628b
commit d9914d37a0
2 changed files with 40 additions and 5 deletions
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28
plugins/ladspa_effect/ladspa_effect.cpp
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@@ -380,6 +380,18 @@ bool ladspaEffect::processAudioBuffer( sampleFrame * _buf,
return( FALSE );
}
sampleFrame * o_buf = NULL;
int frames = _frames;
if( publicName().contains( "C* AmpVTS" ) &&
engine::getMixer()->processingSampleRate() > 88200 )
{
o_buf = _buf;
_buf = new sampleFrame[_frames];
sampleDown( o_buf, _buf, 88200 );
frames = _frames * 88200 /
engine::getMixer()->processingSampleRate();
}
// Copy the LMMS audio buffer to the LADSPA input buffer and initialize
// the control ports. Need to change this to handle non-in-place-broken
// plugins--would speed things up to use the same buffer for both
@@ -393,7 +405,7 @@ bool ladspaEffect::processAudioBuffer( sampleFrame * _buf,
{
case CHANNEL_IN:
for( fpp_t frame = 0;
frame < _frames; frame++ )
frame < frames; frame++ )
{
m_ports[proc][port]->buffer[frame] =
_buf[frame][channel];
@@ -409,7 +421,7 @@ bool ladspaEffect::processAudioBuffer( sampleFrame * _buf,
// treated as though they were control rate by setting the
// port buffer to all the same value.
for( fpp_t frame = 0;
frame < _frames; frame++ )
frame < frames; frame++ )
{
m_ports[proc][port]->buffer[frame] =
m_ports[proc][port]->value;
@@ -441,7 +453,7 @@ bool ladspaEffect::processAudioBuffer( sampleFrame * _buf,
// Process the buffers.
for( ch_cnt_t proc = 0; proc < getProcessorCount(); proc++ )
{
(m_descriptor->run)(m_handles[proc], _frames);
(m_descriptor->run)(m_handles[proc], frames);
}
// Copy the LADSPA output buffers to the LMMS buffer.
@@ -461,7 +473,7 @@ bool ladspaEffect::processAudioBuffer( sampleFrame * _buf,
break;
case CHANNEL_OUT:
for( fpp_t frame = 0;
frame < _frames; frame++ )
frame < frames; frame++ )
{
_buf[frame][channel] =
d *
@@ -483,9 +495,15 @@ bool ladspaEffect::processAudioBuffer( sampleFrame * _buf,
}
}
if( o_buf != NULL )
{
sampleBack( _buf, o_buf, 88200 );
delete[] _buf;
}
// Check whether we need to continue processing input. Restart the
// counter if the threshold has been exceeded.
if( out_sum / _frames <= getGate()+0.000001 )
if( out_sum / frames <= getGate()+0.000001 )
{
incrementBufferCount();
if( getBufferCount() > getTimeout() )