diff --git a/ChangeLog b/ChangeLog
index 72e7c8b83..91f1241f9 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -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
 
diff --git a/include/mixer.h b/include/mixer.h
index 31f86c3a0..ecc4717cd 100644
--- a/include/mixer.h
+++ b/include/mixer.h
@@ -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:
diff --git a/plugins/bass_booster/bassbooster_controls.cpp b/plugins/bass_booster/bassbooster_controls.cpp
index 1635c776f..1abc1525e 100644
--- a/plugins/bass_booster/bassbooster_controls.cpp
+++ b/plugins/bass_booster/bassbooster_controls.cpp
@@ -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 );
 }
diff --git a/plugins/sf2_player/sf2_player.cpp b/plugins/sf2_player/sf2_player.cpp
index 12959e594..28bde1344 100644
--- a/plugins/sf2_player/sf2_player.cpp
+++ b/plugins/sf2_player/sf2_player.cpp
@@ -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;
diff --git a/src/core/audio/audio_device.cpp b/src/core/audio/audio_device.cpp
index 95b77d3f5..00d564c3d 100644
--- a/src/core/audio/audio_device.cpp
+++ b/src/core/audio/audio_device.cpp
@@ -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,