diff --git a/dumb/CMakeLists.txt b/dumb/CMakeLists.txt
index c2cbe8d1b..c35dc7da0 100644
--- a/dumb/CMakeLists.txt
+++ b/dumb/CMakeLists.txt
@@ -36,6 +36,7 @@ add_library( dumb
     src/core/rendsig.c
     src/core/unload.c
     src/helpers/barray.c
+    src/helpers/blip_buf.c
     src/helpers/clickrem.c
     src/helpers/memfile.c
     src/helpers/resample.c
diff --git a/dumb/include/dumb.h b/dumb/include/dumb.h
index 09041ce3c..0b42b4d6d 100644
--- a/dumb/include/dumb.h
+++ b/dumb/include/dumb.h
@@ -662,6 +662,8 @@ typedef struct DUMB_VOLUME_RAMP_INFO DUMB_VOLUME_RAMP_INFO;
 
 typedef void (*DUMB_RESAMPLE_PICKUP)(DUMB_RESAMPLER *resampler, void *data);
 
+#include "internal/blip_buf.h"
+
 struct DUMB_RESAMPLER
 {
 	void *src;
@@ -679,6 +681,9 @@ struct DUMB_RESAMPLER
 		signed char x8[3*2];
 	} x;
 	int overshot;
+	int last_clock;
+	int last_amp[2];
+	blip_t* blip_buffer[2];
 };
 
 struct DUMB_VOLUME_RAMP_INFO
diff --git a/dumb/include/internal/blip_buf.h b/dumb/include/internal/blip_buf.h
new file mode 100644
index 000000000..7f687c3e9
--- /dev/null
+++ b/dumb/include/internal/blip_buf.h
@@ -0,0 +1,77 @@
+/** \file
+Sample buffer that resamples from input clock rate to output sample rate */
+
+/* blip_buf 1.1.0 */
+#ifndef BLIP_BUF_H 
+#define BLIP_BUF_H
+
+#ifdef __cplusplus
+	extern "C" {
+#endif
+
+/** First parameter of most functions is blip_t*, or const blip_t* if nothing
+is changed. */
+typedef struct blip_t blip_t;
+
+/** Creates new buffer that can hold at most sample_count samples. Sets rates
+so that there are blip_max_ratio clocks per sample. Returns pointer to new
+buffer, or NULL if insufficient memory. */
+blip_t* blip_new( int sample_count );
+
+blip_t* blip_dup( blip_t* );
+
+/** Sets approximate input clock rate and output sample rate. For every
+clock_rate input clocks, approximately sample_rate samples are generated. */
+void blip_set_rates( blip_t*, double clock_rate, double sample_rate );
+
+enum { /** Maximum clock_rate/sample_rate ratio. For a given sample_rate,
+clock_rate must not be greater than sample_rate*blip_max_ratio. */
+blip_max_ratio = 1 << 20 };
+
+/** Clears entire buffer. Afterwards, blip_samples_avail() == 0. */
+void blip_clear( blip_t* );
+
+/** Adds positive/negative delta into buffer at specified clock time. */
+void blip_add_delta( blip_t*, unsigned int clock_time, int delta );
+
+/** Same as blip_add_delta(), but uses faster, lower-quality synthesis. */
+void blip_add_delta_fast( blip_t*, unsigned int clock_time, int delta );
+
+/** Length of time frame, in clocks, needed to make sample_count additional
+samples available. */
+int blip_clocks_needed( const blip_t*, int sample_count );
+
+enum { /** Maximum number of samples that can be generated from one time frame. */
+blip_max_frame = 4000 };
+
+/** Makes input clocks before clock_duration available for reading as output
+samples. Also begins new time frame at clock_duration, so that clock time 0 in
+the new time frame specifies the same clock as clock_duration in the old time
+frame specified. Deltas can have been added slightly past clock_duration (up to
+however many clocks there are in two output samples). */
+void blip_end_frame( blip_t*, unsigned int clock_duration );
+
+/** Number of buffered samples available for reading. */
+int blip_samples_avail( const blip_t* );
+
+/** Reads and removes at most 'count' samples and writes them to 'out'. If
+'stereo' is true, writes output to every other element of 'out', allowing easy
+interleaving of two buffers into a stereo sample stream. Outputs 16-bit signed
+samples. Returns number of samples actually read.  */
+int blip_read_samples( blip_t*, int out [], int count );
+
+/** Reads the current integrator and returns it */
+int blip_peek_sample( blip_t* );
+
+/** Frees buffer. No effect if NULL is passed. */
+void blip_delete( blip_t* );
+
+
+/* Deprecated */
+typedef blip_t blip_buffer_t;
+
+#ifdef __cplusplus
+	}
+#endif
+
+#endif
diff --git a/dumb/src/helpers/blip_buf.c b/dumb/src/helpers/blip_buf.c
new file mode 100644
index 000000000..03c111198
--- /dev/null
+++ b/dumb/src/helpers/blip_buf.c
@@ -0,0 +1,354 @@
+/* blip_buf 1.1.0. http://www.slack.net/~ant/ */
+
+#include "internal/blip_buf.h"
+
+#include <assert.h>
+#include <limits.h>
+#include <string.h>
+#include <stdlib.h>
+
+/* Library Copyright (C) 2003-2009 Shay Green. This library is free software;
+you can redistribute it and/or modify it under the terms of the GNU Lesser
+General Public License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version. This
+library is distributed in the hope that it will be useful, but WITHOUT ANY
+WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for more
+details. You should have received a copy of the GNU Lesser General Public
+License along with this module; if not, write to the Free Software Foundation,
+Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */
+
+#if defined (BLARGG_TEST) && BLARGG_TEST
+	#include "blargg_test.h"
+#endif
+
+/* Equivalent to ULONG_MAX >= 0xFFFFFFFF00000000.
+Avoids constants that don't fit in 32 bits. */
+#if ULONG_MAX/0xFFFFFFFF > 0xFFFFFFFF
+	typedef unsigned long fixed_t;
+	enum { pre_shift = 32 };
+
+#elif defined(ULLONG_MAX)
+	typedef unsigned long long fixed_t;
+	enum { pre_shift = 32 };
+
+#else
+	typedef unsigned fixed_t;
+	enum { pre_shift = 0 };
+
+#endif
+
+enum { time_bits = pre_shift + 20 };
+
+static fixed_t const time_unit = (fixed_t) 1 << time_bits;
+
+enum { bass_shift  = 9 }; /* affects high-pass filter breakpoint frequency */
+enum { end_frame_extra = 2 }; /* allows deltas slightly after frame length */
+
+enum { half_width  = 8 };
+enum { buf_extra   = half_width*2 + end_frame_extra };
+enum { phase_bits  = 5 };
+enum { phase_count = 1 << phase_bits };
+enum { delta_bits  = 15 };
+enum { delta_unit  = 1 << delta_bits };
+enum { frac_bits = time_bits - pre_shift };
+
+/* We could eliminate avail and encode whole samples in offset, but that would
+limit the total buffered samples to blip_max_frame. That could only be
+increased by decreasing time_bits, which would reduce resample ratio accuracy.
+*/
+
+/** Sample buffer that resamples to output rate and accumulates samples
+until they're read out */
+struct blip_t
+{
+	fixed_t factor;
+	fixed_t offset;
+	int avail;
+	int size;
+	int integrator;
+};
+
+typedef int buf_t;
+
+/* probably not totally portable */
+#define SAMPLES( buf ) ((buf_t*) ((buf) + 1))
+
+/* Arithmetic (sign-preserving) right shift */
+#define ARITH_SHIFT( n, shift ) \
+	((n) >> (shift))
+
+enum { max_sample = +32767 };
+enum { min_sample = -32768 };
+
+#define CLAMP( n ) \
+	{\
+		if ( (short) n != n )\
+			n = ARITH_SHIFT( n, 16 ) ^ max_sample;\
+	}
+
+static void check_assumptions( void )
+{
+	int n;
+	
+	#if INT_MAX < 0x7FFFFFFF || UINT_MAX < 0xFFFFFFFF
+		#error "int must be at least 32 bits"
+	#endif
+	
+	assert( (-3 >> 1) == -2 ); /* right shift must preserve sign */
+	
+	n = max_sample * 2;
+	CLAMP( n );
+	assert( n == max_sample );
+	
+	n = min_sample * 2;
+	CLAMP( n );
+	assert( n == min_sample );
+	
+	assert( blip_max_ratio <= time_unit );
+	assert( blip_max_frame <= (fixed_t) -1 >> time_bits );
+}
+
+blip_t* blip_new( int size )
+{
+	blip_t* m;
+	assert( size >= 0 );
+	
+	m = (blip_t*) malloc( sizeof *m + (size + buf_extra) * sizeof (buf_t) );
+	if ( m )
+	{
+		m->factor = time_unit / blip_max_ratio;
+		m->size   = size;
+		blip_clear( m );
+		check_assumptions();
+	}
+	return m;
+}
+
+blip_t* blip_dup( blip_t* m )
+{
+	size_t size = sizeof *m + (m->size + buf_extra) * sizeof(buf_t);
+	blip_t* r = (blip_t*) malloc( size );
+	if ( r ) memcpy( r, m, size );
+	return r;
+}
+
+void blip_delete( blip_t* m )
+{
+	if ( m != NULL )
+	{
+		/* Clear fields in case user tries to use after freeing */
+		memset( m, 0, sizeof *m );
+		free( m );
+	}
+}
+
+void blip_set_rates( blip_t* m, double clock_rate, double sample_rate )
+{
+	double factor = time_unit * sample_rate / clock_rate;
+	m->factor = (fixed_t) factor;
+	
+	/* Fails if clock_rate exceeds maximum, relative to sample_rate */
+	assert( 0 <= factor - m->factor && factor - m->factor < 1 );
+	
+	/* Avoid requiring math.h. Equivalent to
+	m->factor = (int) ceil( factor ) */
+	if ( m->factor < factor )
+		m->factor++;
+	
+	/* At this point, factor is most likely rounded up, but could still
+	have been rounded down in the floating-point calculation. */
+}
+
+void blip_clear( blip_t* m )
+{
+	/* We could set offset to 0, factor/2, or factor-1. 0 is suitable if
+	factor is rounded up. factor-1 is suitable if factor is rounded down.
+	Since we don't know rounding direction, factor/2 accommodates either,
+	with the slight loss of showing an error in half the time. Since for
+	a 64-bit factor this is years, the halving isn't a problem. */
+	
+	m->offset     = m->factor / 2;
+	m->avail      = 0;
+	m->integrator = 0;
+	memset( SAMPLES( m ), 0, (m->size + buf_extra) * sizeof (buf_t) );
+}
+
+int blip_clocks_needed( const blip_t* m, int samples )
+{
+	fixed_t needed;
+	
+	/* Fails if buffer can't hold that many more samples */
+	assert( samples >= 0 && m->avail + samples <= m->size );
+	
+	needed = (fixed_t) samples * time_unit;
+	if ( needed < m->offset )
+		return 0;
+	
+	return (int)((needed - m->offset + m->factor - 1) / m->factor);
+}
+
+void blip_end_frame( blip_t* m, unsigned t )
+{
+	fixed_t off = t * m->factor + m->offset;
+	m->avail += (int)(off >> time_bits);
+	m->offset = off & (time_unit - 1);
+	
+	/* Fails if buffer size was exceeded */
+	assert( m->avail <= m->size );
+}
+
+int blip_samples_avail( const blip_t* m )
+{
+	return m->avail;
+}
+
+static void remove_samples( blip_t* m, int count )
+{
+	buf_t* buf = SAMPLES( m );
+	int remain = m->avail + buf_extra - count;
+	m->avail -= count;
+	
+	memmove( &buf [0], &buf [count], remain * sizeof buf [0] );
+	memset( &buf [remain], 0, count * sizeof buf [0] );
+}
+
+int blip_read_samples( blip_t* m, int out [], int count )
+{
+	assert( count >= 0 );
+	
+	if ( count > m->avail )
+		count = m->avail;
+	
+	if ( count )
+	{
+		buf_t const* in  = SAMPLES( m );
+		buf_t const* end = in + count;
+		int sum = m->integrator;
+		do
+		{
+			/* Eliminate fraction */
+			int s = ARITH_SHIFT( sum, delta_bits - 8 );
+			
+			sum += *in++;
+			
+			*out = s;
+			out++;
+			
+			/* High-pass filter */
+			sum -= s >> (8 - (delta_bits - bass_shift)); //<< (delta_bits - bass_shift - 8);
+		}
+		while ( in != end );
+		m->integrator = sum;
+		
+		remove_samples( m, count );
+	}
+	
+	return count;
+}
+
+int blip_peek_sample( blip_t* m )
+{
+	return ARITH_SHIFT( m->integrator, delta_bits - 8 );
+}
+
+/* Things that didn't help performance on x86:
+	__attribute__((aligned(128)))
+	#define short int
+	restrict
+*/
+
+/* Sinc_Generator( 0.9, 0.55, 4.5 ) */
+static short const bl_step [phase_count + 1] [half_width] =
+{
+{   43, -115,  350, -488, 1136, -914, 5861,21022},
+{   44, -118,  348, -473, 1076, -799, 5274,21001},
+{   45, -121,  344, -454, 1011, -677, 4706,20936},
+{   46, -122,  336, -431,  942, -549, 4156,20829},
+{   47, -123,  327, -404,  868, -418, 3629,20679},
+{   47, -122,  316, -375,  792, -285, 3124,20488},
+{   47, -120,  303, -344,  714, -151, 2644,20256},
+{   46, -117,  289, -310,  634,  -17, 2188,19985},
+{   46, -114,  273, -275,  553,  117, 1758,19675},
+{   44, -108,  255, -237,  471,  247, 1356,19327},
+{   43, -103,  237, -199,  390,  373,  981,18944},
+{   42,  -98,  218, -160,  310,  495,  633,18527},
+{   40,  -91,  198, -121,  231,  611,  314,18078},
+{   38,  -84,  178,  -81,  153,  722,   22,17599},
+{   36,  -76,  157,  -43,   80,  824, -241,17092},
+{   34,  -68,  135,   -3,    8,  919, -476,16558},
+{   32,  -61,  115,   34,  -60, 1006, -683,16001},
+{   29,  -52,   94,   70, -123, 1083, -862,15422},
+{   27,  -44,   73,  106, -184, 1152,-1015,14824},
+{   25,  -36,   53,  139, -239, 1211,-1142,14210},
+{   22,  -27,   34,  170, -290, 1261,-1244,13582},
+{   20,  -20,   16,  199, -335, 1301,-1322,12942},
+{   18,  -12,   -3,  226, -375, 1331,-1376,12293},
+{   15,   -4,  -19,  250, -410, 1351,-1408,11638},
+{   13,    3,  -35,  272, -439, 1361,-1419,10979},
+{   11,    9,  -49,  292, -464, 1362,-1410,10319},
+{    9,   16,  -63,  309, -483, 1354,-1383, 9660},
+{    7,   22,  -75,  322, -496, 1337,-1339, 9005},
+{    6,   26,  -85,  333, -504, 1312,-1280, 8355},
+{    4,   31,  -94,  341, -507, 1278,-1205, 7713},
+{    3,   35, -102,  347, -506, 1238,-1119, 7082},
+{    1,   40, -110,  350, -499, 1190,-1021, 6464},
+{    0,   43, -115,  350, -488, 1136, -914, 5861}
+};
+
+/* Shifting by pre_shift allows calculation using unsigned int rather than
+possibly-wider fixed_t. On 32-bit platforms, this is likely more efficient.
+And by having pre_shift 32, a 32-bit platform can easily do the shift by
+simply ignoring the low half. */
+
+void blip_add_delta( blip_t* m, unsigned time, int delta )
+{
+	unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
+	buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits);
+	
+	int const phase_shift = frac_bits - phase_bits;
+	int phase = fixed >> phase_shift & (phase_count - 1);
+	short const* in  = bl_step [phase];
+	short const* rev = bl_step [phase_count - phase];
+	
+	int interp = fixed >> (phase_shift - delta_bits) & (delta_unit - 1);
+	int delta2 = (delta * interp) >> delta_bits;
+	delta -= delta2;
+	
+	/* Fails if buffer size was exceeded */
+	assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] );
+	
+	out [0] += in[0]*delta + in[half_width+0]*delta2;
+	out [1] += in[1]*delta + in[half_width+1]*delta2;
+	out [2] += in[2]*delta + in[half_width+2]*delta2;
+	out [3] += in[3]*delta + in[half_width+3]*delta2;
+	out [4] += in[4]*delta + in[half_width+4]*delta2;
+	out [5] += in[5]*delta + in[half_width+5]*delta2;
+	out [6] += in[6]*delta + in[half_width+6]*delta2;
+	out [7] += in[7]*delta + in[half_width+7]*delta2;
+	
+	in = rev;
+	out [ 8] += in[7]*delta + in[7-half_width]*delta2;
+	out [ 9] += in[6]*delta + in[6-half_width]*delta2;
+	out [10] += in[5]*delta + in[5-half_width]*delta2;
+	out [11] += in[4]*delta + in[4-half_width]*delta2;
+	out [12] += in[3]*delta + in[3-half_width]*delta2;
+	out [13] += in[2]*delta + in[2-half_width]*delta2;
+	out [14] += in[1]*delta + in[1-half_width]*delta2;
+	out [15] += in[0]*delta + in[0-half_width]*delta2;
+}
+
+void blip_add_delta_fast( blip_t* m, unsigned time, int delta )
+{
+	unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
+	buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits);
+	
+	int interp = fixed >> (frac_bits - delta_bits) & (delta_unit - 1);
+	int delta2 = delta * interp;
+	
+	/* Fails if buffer size was exceeded */
+	assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] );
+	
+	out [7] += delta * delta_unit - delta2;
+	out [8] += delta2;
+}
diff --git a/dumb/src/helpers/resamp2.inc b/dumb/src/helpers/resamp2.inc
index 84826ce51..de65dfb34 100644
--- a/dumb/src/helpers/resamp2.inc
+++ b/dumb/src/helpers/resamp2.inc
@@ -95,7 +95,8 @@ static int process_pickup(DUMB_RESAMPLER *resampler)
 #define SET_VOLUME_VARIABLES SET_MONO_DEST_VOLUME_VARIABLES
 #define RETURN_VOLUME_VARIABLES RETURN_MONO_DEST_VOLUME_VARIABLES
 #define VOLUMES_ARE_ZERO MONO_DEST_VOLUMES_ARE_ZERO
-#define MIX_ALIAS(op, upd, offset) MONO_DEST_MIX_ALIAS(op, upd, offset)
+#define MIX_ALIAS(count) MONO_DEST_MIX_ALIAS(count)
+#define PEEK_ALIAS MONO_DEST_PEEK_ALIAS
 #define MIX_LINEAR(op, upd, o0, o1) MONO_DEST_MIX_LINEAR(op, upd, o0, o1)
 #define MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3) MONO_DEST_MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3)
 #define MIX_ZEROS(op) *dst++ op 0
@@ -137,7 +138,8 @@ static int process_pickup(DUMB_RESAMPLER *resampler)
 	if ( volume_right ) volume_right->volume = (float)rvolr / 16777216.0f; \
 }
 #define VOLUMES_ARE_ZERO (lvol == 0 && lvolt == 0 && rvol == 0 && rvolt == 0)
-#define MIX_ALIAS(op, upd, offset) STEREO_DEST_MIX_ALIAS(op, upd, offset)
+#define MIX_ALIAS(count) STEREO_DEST_MIX_ALIAS(count)
+#define PEEK_ALIAS STEREO_DEST_PEEK_ALIAS
 #define MIX_LINEAR(op, upd, o0, o1) STEREO_DEST_MIX_LINEAR(op, upd, o0, o1)
 #define MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3) STEREO_DEST_MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3)
 #define MIX_ZEROS(op) { *dst++ op 0; *dst++ op 0; }
@@ -157,6 +159,9 @@ static int process_pickup(DUMB_RESAMPLER *resampler)
 #undef MONO_DEST_VOLUME_ZEROS
 #undef MONO_DEST_VOLUME_VARIABLES
 #undef MONO_DEST_VOLUME_PARAMETERS
+#undef STEREO_DEST_PEEK_ALIAS
+#undef MONO_DEST_PEEK_ALIAS
+#undef POKE_ALIAS
 #undef COPYSRC2
 #undef COPYSRC
 #undef DIVIDE_BY_SRC_CHANNELS
diff --git a/dumb/src/helpers/resamp3.inc b/dumb/src/helpers/resamp3.inc
index 3d61bfff1..1581e7bfa 100644
--- a/dumb/src/helpers/resamp3.inc
+++ b/dumb/src/helpers/resamp3.inc
@@ -46,12 +46,13 @@
 
 int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VOLUME_PARAMETERS, double delta)
 {
-	int dt;
+	int dt, inv_dt;
 	int VOLUME_VARIABLES;
 	long done;
 	long todo;
 	LONG_LONG todo64;
 	int quality;
+	int blip_samples[256*SRC_CHANNELS];
 
 	if (!resampler || resampler->dir == 0) return 0;
 	ASSERT(resampler->dir == -1 || resampler->dir == 1);
@@ -59,6 +60,7 @@ int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VO
 	done = 0;
 	dt = (int)(delta * 65536.0 + 0.5);
 	if (dt == 0 || dt == 0x80000000) return 0;
+	inv_dt = (int)(1.0 / delta * 65536.0 + 0.5);
 	SET_VOLUME_VARIABLES;
 
 	if (VOLUMES_ARE_ZERO) dst = NULL;
@@ -104,29 +106,34 @@ int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VO
 					subpos = (long)new_subpos & 65535;
 				} else if (quality <= DUMB_RQ_ALIASING) {
 					/* Aliasing, backwards */
-					SRCTYPE xbuf[2*SRC_CHANNELS];
+					int todo_clocks = todo << 16, todo_clocks_set = todo_clocks;
+ 					SRCTYPE xbuf[2*SRC_CHANNELS];
 					SRCTYPE *x = &xbuf[0];
-					SRCTYPE *xstart;
 					COPYSRC(xbuf, 0, resampler->X, 1);
 					COPYSRC(xbuf, 1, resampler->X, 2);
-					while (todo && x < &xbuf[2*SRC_CHANNELS]) {
+					if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
+					while (resampler->last_clock < todo_clocks_set && x < &xbuf[2*SRC_CHANNELS]) {
 						// TODO: check what happens when multiple tempo slides occur per row
 						HEAVYASSERT(pos >= resampler->start);
-						MIX_ALIAS(+=, 1, 0);
-						subpos += dt;
-						pos += subpos >> 16;
-						x -= (subpos >> 16) * SRC_CHANNELS;
-						subpos &= 65535;
-						todo--;
+						POKE_ALIAS(0);
+						pos--;
+						x += SRC_CHANNELS;
+					}
+					x = &src[pos*SRC_CHANNELS];
+					while ( todo_clocks ) {
+						todo_clocks_set = todo_clocks;
+						if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
+						todo_clocks -= todo_clocks_set;
+						todo = ( todo_clocks_set - resampler->last_clock + inv_dt - 1 ) / inv_dt;
+						if ( todo < 0 ) todo = 0;
+						LOOP4(todo,
+							POKE_ALIAS(2);
+							pos--;
+							x -= SRC_CHANNELS;
+						);
+						todo = todo_clocks_set >> 16;
+						MIX_ALIAS( todo );
 					}
-					x = xstart = &src[pos*SRC_CHANNELS];
-					LOOP4(todo,
-						MIX_ALIAS(+=, 1, 2);
-						subpos += dt;
-						x += (subpos >> 16) * SRC_CHANNELS;
-						subpos &= 65535;
-					);
-					pos += DIVIDE_BY_SRC_CHANNELS((long)(x - xstart));
 				} else if (quality <= DUMB_RQ_LINEAR) {
 					/* Linear interpolation, backwards */
 					SRCTYPE xbuf[3*SRC_CHANNELS];
@@ -205,28 +212,33 @@ int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VO
 					subpos = (long)new_subpos & 65535;
 				} else if (quality <= DUMB_RQ_ALIASING) {
 					/* Aliasing, forwards */
+					int todo_clocks = todo << 16, todo_clocks_set = todo_clocks;
 					SRCTYPE xbuf[2*SRC_CHANNELS];
 					SRCTYPE *x = &xbuf[0];
-					SRCTYPE *xstart;
 					COPYSRC(xbuf, 0, resampler->X, 1);
 					COPYSRC(xbuf, 1, resampler->X, 2);
-					while (todo && x < &xbuf[2*SRC_CHANNELS]) {
+					if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
+					while (resampler->last_clock < todo_clocks_set && x < &xbuf[2*SRC_CHANNELS]) {
 						HEAVYASSERT(pos < resampler->end);
-						MIX_ALIAS(+=, 1, 0);
-						subpos += dt;
-						pos += subpos >> 16;
-						x += (subpos >> 16) * SRC_CHANNELS;
-						subpos &= 65535;
-						todo--;
+						POKE_ALIAS(0);
+						pos++;
+						x += SRC_CHANNELS;
+					}
+					x = &src[pos*SRC_CHANNELS];
+					while ( todo_clocks ) {
+						todo_clocks_set = todo_clocks;
+						if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
+						todo_clocks -= todo_clocks_set;
+						todo = ( todo_clocks_set - resampler->last_clock + inv_dt - 1 ) / inv_dt;
+						if ( todo < 0 ) todo = 0;
+						LOOP4(todo,
+							POKE_ALIAS(-2);
+							pos++;
+							x += SRC_CHANNELS;
+						);
+						todo = todo_clocks_set >> 16;
+						MIX_ALIAS( todo );
 					}
-					x = xstart = &src[pos*SRC_CHANNELS];
-					LOOP4(todo,
-						MIX_ALIAS(+=, 1, -2);
-						subpos += dt;
-						x += (subpos >> 16) * SRC_CHANNELS;
-						subpos &= 65535;
-					);
-					pos += DIVIDE_BY_SRC_CHANNELS((long)(x - xstart));
 				} else if (quality <= DUMB_RQ_LINEAR) {
 					/* Linear interpolation, forwards */
 					SRCTYPE xbuf[3*SRC_CHANNELS];
@@ -339,7 +351,7 @@ void dumb_resample_get_current_sample(DUMB_RESAMPLER *resampler, VOLUME_PARAMETE
 		HEAVYASSERT(pos >= resampler->start);
 		if (dumb_resampling_quality <= DUMB_RQ_ALIASING) {
 			/* Aliasing, backwards */
-			MIX_ALIAS(=, 0, 1);
+			PEEK_ALIAS;
 		} else if (quality <= DUMB_RQ_LINEAR) {
 			/* Linear interpolation, backwards */
 			MIX_LINEAR(=, 0, 2, 1);
@@ -351,7 +363,7 @@ void dumb_resample_get_current_sample(DUMB_RESAMPLER *resampler, VOLUME_PARAMETE
 		HEAVYASSERT(pos < resampler->end);
 		if (dumb_resampling_quality <= DUMB_RQ_ALIASING) {
 			/* Aliasing */
-			MIX_ALIAS(=, 0, 1);
+			PEEK_ALIAS;
 		} else if (dumb_resampling_quality <= DUMB_RQ_LINEAR) {
 			/* Linear interpolation, forwards */
 			MIX_LINEAR(=, 0, 1, 2);
@@ -368,6 +380,7 @@ void dumb_resample_get_current_sample(DUMB_RESAMPLER *resampler, VOLUME_PARAMETE
 #undef MIX_CUBIC
 #undef MIX_LINEAR
 #undef MIX_ALIAS
+#undef PEEK_ALIAS
 #undef VOLUMES_ARE_ZERO
 #undef SET_VOLUME_VARIABLES
 #undef RETURN_VOLUME_VARIABLES
diff --git a/dumb/src/helpers/resample.c b/dumb/src/helpers/resample.c
index db42ee7ae..e78d0d39c 100644
--- a/dumb/src/helpers/resample.c
+++ b/dumb/src/helpers/resample.c
@@ -189,7 +189,7 @@ static void init_cubic(void)
 
 #define SRCTYPE sample_t
 #define SRCBITS 24
-#define ALIAS(x, vol) MULSC(x, vol)
+#define ALIAS(x) (x >> 8)
 #define LINEAR(x0, x1) (x0 + MULSC(x1 - x0, subpos))
 /*
 #define SET_CUBIC_COEFFICIENTS(x0, x1, x2, x3) { \
@@ -225,7 +225,7 @@ static void init_cubic(void)
 #define SUFFIX _16
 #define SRCTYPE short
 #define SRCBITS 16
-#define ALIAS(x, vol) (x * vol >> 8)
+#define ALIAS(x) (x)
 #define LINEAR(x0, x1) ((x0 << 8) + MULSC16(x1 - x0, subpos))
 /*
 #define SET_CUBIC_COEFFICIENTS(x0, x1, x2, x3) { \
@@ -247,7 +247,7 @@ static void init_cubic(void)
 #define SUFFIX _8
 #define SRCTYPE signed char
 #define SRCBITS 8
-#define ALIAS(x, vol) (x * vol)
+#define ALIAS(x) (x << 8)
 #define LINEAR(x0, x1) ((x0 << 16) + (x1 - x0) * subpos)
 /*
 #define SET_CUBIC_COEFFICIENTS(x0, x1, x2, x3) { \
diff --git a/dumb/src/helpers/resample.inc b/dumb/src/helpers/resample.inc
index eaef2976c..364e36ccd 100644
--- a/dumb/src/helpers/resample.inc
+++ b/dumb/src/helpers/resample.inc
@@ -69,6 +69,11 @@ void dumb_reset_resampler(DUMB_RESAMPLER *resampler, SRCTYPE *src, int src_chann
 	}
 	for (i = 0; i < src_channels*3; i++) resampler->X[i] = 0;
 	resampler->overshot = -1;
+	resampler->last_clock = 0;
+	resampler->last_amp[0] = 0;
+	resampler->last_amp[1] = 0;
+	blip_clear(resampler->blip_buffer[0]);
+	blip_clear(resampler->blip_buffer[1]);
 }
 
 
@@ -77,6 +82,21 @@ DUMB_RESAMPLER *dumb_start_resampler(SRCTYPE *src, int src_channels, int32 pos,
 {
 	DUMB_RESAMPLER *resampler = malloc(sizeof(*resampler));
 	if (!resampler) return NULL;
+	resampler->blip_buffer[0] = blip_new( 256 );
+	if (!resampler->blip_buffer[0])
+	{
+		free(resampler);
+		return NULL;
+	}
+	resampler->blip_buffer[1] = blip_new( 256 );
+	if (!resampler->blip_buffer[1])
+	{
+		free(resampler->blip_buffer[0]);
+		free(resampler);
+		return NULL;
+	}
+	blip_set_rates(resampler->blip_buffer[0], 65536, 1);
+	blip_set_rates(resampler->blip_buffer[1], 65536, 1);
 	dumb_reset_resampler(resampler, src, src_channels, pos, start, end, quality);
 	return resampler;
 }
@@ -123,16 +143,41 @@ DUMB_RESAMPLER *dumb_start_resampler(SRCTYPE *src, int src_channels, int32 pos,
 }
 #define RETURN_MONO_DEST_VOLUME_VARIABLES if ( volume ) volume->volume = (float)volr / 16777216.0f
 #define MONO_DEST_VOLUMES_ARE_ZERO (vol == 0 && volt == 0)
-#define MONO_DEST_MIX_ALIAS(op, upd, offset) { \
-	*dst++ op ALIAS(x[offset], vol); \
-	if ( upd ) UPDATE_VOLUME( volume, vol ); \
+#define POKE_ALIAS(offset) { \
+	int delta = ALIAS(x[offset]) - resampler->last_amp[0]; \
+	resampler->last_amp[0] += delta; \
+	if ( delta ) blip_add_delta( resampler->blip_buffer[0], resampler->last_clock, delta ); \
+	resampler->last_clock += inv_dt; \
 }
-#define STEREO_DEST_MIX_ALIAS(op, upd, offset) { \
-	int xm = x[offset]; \
-	*dst++ op ALIAS(xm, lvol); \
-	*dst++ op ALIAS(xm, rvol); \
-	if ( upd ) UPDATE_VOLUME( volume_left, lvol ); \
-	if ( upd ) UPDATE_VOLUME( volume_right, rvol ); \
+#define MONO_DEST_PEEK_ALIAS *dst = MULSC( blip_peek_sample( resampler->blip_buffer[0] ), vol )
+#define MONO_DEST_MIX_ALIAS(count) { \
+	int n = 0; \
+	resampler->last_clock -= count * 65536; \
+	blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
+	blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
+	LOOP4( count, \
+		*dst++ += MULSC( blip_samples[n], vol ); \
+		n++; \
+		UPDATE_VOLUME( volume, vol ); \
+	); \
+}
+#define STEREO_DEST_PEEK_ALIAS { \
+	int sample = blip_peek_sample( resampler->blip_buffer[0] ); \
+	*dst++ = MULSC( sample, lvol ); \
+	*dst++ = MULSC( sample, rvol ); \
+}
+#define STEREO_DEST_MIX_ALIAS(count) { \
+	int sample, n = 0; \
+	resampler->last_clock -= count * 65536; \
+	blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
+	blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
+	LOOP4( count, \
+		sample = blip_samples[n++]; \
+		*dst++ += MULSC( sample, lvol ); \
+		*dst++ += MULSC( sample, rvol ); \
+		UPDATE_VOLUME( volume_left, lvol ); \
+		UPDATE_VOLUME( volume_right, rvol ); \
+	); \
 }
 #define MONO_DEST_MIX_LINEAR(op, upd, o0, o1) { \
 	*dst++ op MULSC(LINEAR(x[o0], x[o1]), vol); \
@@ -208,16 +253,51 @@ DUMB_RESAMPLER *dumb_start_resampler(SRCTYPE *src, int src_channels, int32 pos,
 	if ( volume_right ) volume_right->volume = (float)rvolr / 16777216.0f; \
 }
 #define MONO_DEST_VOLUMES_ARE_ZERO (lvol == 0 && lvolt == 0 && rvol == 0 && rvolt == 0)
-#define MONO_DEST_MIX_ALIAS(op, upd, offset) { \
-	*dst++ op ALIAS(x[(offset)*2], lvol) + ALIAS(x[(offset)*2+1], rvol); \
-	if ( upd ) UPDATE_VOLUME( volume_left, lvol ); \
-	if ( upd ) UPDATE_VOLUME( volume_right, rvol ); \
+#define POKE_ALIAS(offset) { \
+	int deltal = ALIAS(x[(offset)*2+0]) - resampler->last_amp[0]; \
+	int deltar = ALIAS(x[(offset)*2+1]) - resampler->last_amp[1]; \
+	resampler->last_amp[0] += deltal; \
+	resampler->last_amp[1] += deltar; \
+	if ( deltal ) blip_add_delta( resampler->blip_buffer[0], resampler->last_clock, deltal ); \
+	if ( deltar ) blip_add_delta( resampler->blip_buffer[1], resampler->last_clock, deltar ); \
+	resampler->last_clock += inv_dt; \
 }
-#define STEREO_DEST_MIX_ALIAS(op, upd, offset) { \
-	*dst++ op ALIAS(x[(offset)*2], lvol); \
-	*dst++ op ALIAS(x[(offset)*2+1], rvol); \
-	if ( upd ) UPDATE_VOLUME( volume_left, lvol ); \
-	if ( upd ) UPDATE_VOLUME( volume_right, rvol ); \
+#define MONO_DEST_PEEK_ALIAS { \
+	*dst = MULSC( blip_peek_sample( resampler->blip_buffer[0] ), lvol ) + \
+		MULSC( blip_peek_sample( resampler->blip_buffer[1] ), rvol ); \
+}
+#define MONO_DEST_MIX_ALIAS(count) { \
+	int n = 0; \
+	resampler->last_clock -= count * 65536; \
+	blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
+	blip_end_frame( resampler->blip_buffer[1], count * 65536 ); \
+	blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
+	blip_read_samples( resampler->blip_buffer[1], blip_samples + 256, count ); \
+	LOOP4( count, \
+		*dst++ += MULSC( blip_samples[n], lvol ) + MULSC( blip_samples[256+n], rvol ); \
+		n++; \
+		UPDATE_VOLUME( volume_left, lvol ); \
+		UPDATE_VOLUME( volume_right, rvol ); \
+	); \
+}
+#define STEREO_DEST_PEEK_ALIAS { \
+	*dst++ = MULSC( blip_peek_sample( resampler->blip_buffer[0] ), lvol ); \
+	*dst++ = MULSC( blip_peek_sample( resampler->blip_buffer[1] ), rvol ); \
+}
+#define STEREO_DEST_MIX_ALIAS(count) { \
+	int n = 0; \
+	resampler->last_clock -= count * 65536; \
+	blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
+	blip_end_frame( resampler->blip_buffer[1], count * 65536 ); \
+	blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
+	blip_read_samples( resampler->blip_buffer[1], blip_samples + 256, count ); \
+	LOOP4( count, \
+		*dst++ += MULSC( blip_samples[n], lvol); \
+		*dst++ += MULSC( blip_samples[256+n], rvol); \
+		n++; \
+		UPDATE_VOLUME( volume_left, lvol ); \
+		UPDATE_VOLUME( volume_right, rvol ); \
+	); \
 }
 #define MONO_DEST_MIX_LINEAR(op, upd, o0, o1) { \
 	*dst++ op MULSC(LINEAR(x[(o0)*2], x[(o1)*2]), lvol) + MULSC(LINEAR(x[(o0)*2+1], x[(o1)*2+1]), rvol); \
diff --git a/dumb/src/it/itrender.c b/dumb/src/it/itrender.c
index 24530156d..4d188ae6a 100644
--- a/dumb/src/it/itrender.c
+++ b/dumb/src/it/itrender.c
@@ -29,15 +29,38 @@
 #define END_RAMPING
 #define RAMP_DOWN
 
-static IT_PLAYING *alloc_playing(DUMB_IT_SIGRENDERER *itsr)
+static IT_PLAYING *new_playing(DUMB_IT_SIGRENDERER *itsr)
 {
+	IT_PLAYING *r;
+
 	if (itsr->free_playing != NULL)
 	{
-		IT_PLAYING *pl = itsr->free_playing;
-		itsr->free_playing = pl->next;
-		return pl;
+		r = itsr->free_playing;
+		itsr->free_playing = r->next;
+		blip_clear(r->resampler.blip_buffer[0]);
+		blip_clear(r->resampler.blip_buffer[1]);
+		return r;
 	}
-	return (IT_PLAYING *)malloc(sizeof(IT_PLAYING));
+	r = (IT_PLAYING *)malloc(sizeof(IT_PLAYING));
+	if (r)
+	{
+		r->resampler.blip_buffer[0] = blip_new( 256 );
+		if ( !r->resampler.blip_buffer[0] )
+		{
+			free( r );
+			return NULL;
+		}
+		r->resampler.blip_buffer[1] = blip_new( 256 );
+		if ( !r->resampler.blip_buffer[1] )
+		{
+			free( r->resampler.blip_buffer[0] );
+			free( r );
+			return NULL;
+		}
+		blip_set_rates(r->resampler.blip_buffer[0], 65536, 1);
+		blip_set_rates(r->resampler.blip_buffer[1], 65536, 1);
+	}
+	return r;
 }
 
 static void free_playing(DUMB_IT_SIGRENDERER *itsr, IT_PLAYING *playing)
@@ -46,6 +69,13 @@ static void free_playing(DUMB_IT_SIGRENDERER *itsr, IT_PLAYING *playing)
 	itsr->free_playing = playing;
 }
 
+static void free_playing_orig(IT_PLAYING * r)
+{
+	blip_delete( r->resampler.blip_buffer[1] );
+	blip_delete( r->resampler.blip_buffer[0] );
+	free( r );
+}
+
 static IT_PLAYING *dup_playing(IT_PLAYING *src, IT_CHANNEL *dstchannel, IT_CHANNEL *srcchannel)
 {
 	IT_PLAYING *dst;
@@ -135,6 +165,19 @@ static IT_PLAYING *dup_playing(IT_PLAYING *src, IT_CHANNEL *dstchannel, IT_CHANN
 
 	dst->resampler = src->resampler;
 	dst->resampler.pickup_data = dst;
+	dst->resampler.blip_buffer[0] = blip_dup( dst->resampler.blip_buffer[0] );
+	if ( !dst->resampler.blip_buffer[0] )
+	{
+		free( dst );
+		return NULL;
+	}
+	dst->resampler.blip_buffer[1] = blip_dup( dst->resampler.blip_buffer[1] );
+	if ( !dst->resampler.blip_buffer[1] )
+	{
+		blip_delete( dst->resampler.blip_buffer[0] );
+		free( dst );
+		return NULL;
+	}
 	dst->time_lost = src->time_lost;
 
 	//dst->output = src->output;
@@ -1582,7 +1625,7 @@ static void it_retrigger_note(DUMB_IT_SIGRENDERER *sigrenderer, IT_CHANNEL *chan
 	if (channel->playing)
 		free_playing(sigrenderer, channel->playing);
 
-	channel->playing = alloc_playing(sigrenderer);
+	channel->playing = new_playing(sigrenderer);
 
 	if (!channel->playing)
 		return;
@@ -2889,9 +2932,7 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 {
 	DUMB_IT_SIGDATA *sigdata = sigrenderer->sigdata;
 	IT_CHANNEL *channel = &sigrenderer->channel[(int)entry->channel];
-	IT_PLAYING playing;
-
-	playing.sample = 0;
+	IT_PLAYING * playing = NULL;
 
 	if (entry->mask & IT_ENTRY_INSTRUMENT) {
 		int oldsample = channel->sample;
@@ -2901,7 +2942,8 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 		if (channel->playing &&
 			!((entry->mask & IT_ENTRY_NOTE) && entry->note >= 120) &&
 			!((entry->mask & IT_ENTRY_EFFECT) && entry->effect == IT_XM_KEY_OFF && entry->effectvalue == 0)) {
-			playing = *channel->playing;
+			playing = dup_playing(channel->playing, channel, channel);
+			if (!playing) return;
 			if (!(sigdata->flags & IT_WAS_A_MOD)) {
 				/* Retrigger vol/pan envelopes if enabled, and cancel fadeout.
 				 * Also reset vol/pan to that of _original_ instrument.
@@ -2934,12 +2976,11 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 							channel->playing = NULL;
 						}
 					} else {
-						if (!channel->playing) {
-							channel->playing = alloc_playing(sigrenderer);
-							if (!channel->playing) return;
+						if (channel->playing) {
+							free_playing(sigrenderer, channel->playing);
 						}
-						*channel->playing = playing;
-						playing.sample = (IT_SAMPLE *)-1;
+						channel->playing = playing;
+						playing = NULL;
 						channel->playing->declick_stage = 0;
 						channel->playing->declick_volume = 1.f / 256.f;
 #else
@@ -2985,7 +3026,11 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 			if (channel->playing) {
 #ifdef RAMP_DOWN
 				int i;
-				if (playing.sample) *channel->playing = playing;
+				if (playing) {
+					free_playing(sigrenderer, channel->playing);
+					channel->playing = playing;
+					playing = NULL;
+				}
 				for (i = 0; i < DUMB_IT_N_NNA_CHANNELS; i++) {
 					if (!sigrenderer->playing[i]) {
 						channel->playing->declick_stage = 2;
@@ -3003,6 +3048,7 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 				channel->playing = NULL;
 #endif
 			}
+			if (playing) free_playing(sigrenderer, playing);
 			return;
 		} else if (channel->playing && (entry->mask & IT_ENTRY_VOLPAN) && ((entry->volpan>>4) == 0xF)) {
 			/* Don't retrigger note; portamento in the volume column. */
@@ -3015,20 +3061,26 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 			channel->destnote = IT_NOTE_OFF;
 
 			if (!channel->playing) {
-				channel->playing = alloc_playing(sigrenderer);
-				if (!channel->playing)
+				channel->playing = new_playing(sigrenderer);
+				if (!channel->playing) {
+					if (playing) free_playing(sigrenderer, playing);
 					return;
+				}
 				// Adding the following seems to do the trick for the case where a piece starts with an instrument alone and then some notes alone.
 				retrigger_xm_envelopes(channel->playing);
 			}
 #ifdef RAMP_DOWN
-			else if (playing.sample != (IT_SAMPLE *)-1) {
+			else if (playing) {
 				/* volume rampy stuff! move note to NNA */
 				int i;
-				IT_PLAYING * ptemp = alloc_playing(sigrenderer);
-				if (!ptemp) return;
-				if (playing.sample) *ptemp = playing;
-				else *ptemp = *channel->playing;
+				IT_PLAYING * ptemp;
+				if (playing->sample) ptemp = playing;
+				else ptemp = channel->playing;
+				if (!ptemp) {
+					if (playing) free_playing(sigrenderer, playing);
+					return;
+				}
+				playing = NULL;
 				for (i = 0; i < DUMB_IT_N_NNA_CHANNELS; i++) {
 					if (!sigrenderer->playing[i]) {
 						ptemp->declick_stage = 2;
@@ -3189,6 +3241,8 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
 				break;
 		}
 	}
+
+	if (playing) free_playing(sigrenderer, playing);
 }
 
 
@@ -5165,7 +5219,7 @@ void _dumb_it_end_sigrenderer(sigrenderer_t *vsigrenderer)
 
 		for (i = 0; i < DUMB_IT_N_CHANNELS; i++) {
 			if (sigrenderer->channel[i].playing)
-				free(sigrenderer->channel[i].playing);
+				free_playing_orig(sigrenderer->channel[i].playing);
 #ifdef BIT_ARRAY_BULLSHIT
 			bit_array_destroy(sigrenderer->channel[i].played_patjump);
 #endif
@@ -5173,12 +5227,12 @@ void _dumb_it_end_sigrenderer(sigrenderer_t *vsigrenderer)
 
 		for (i = 0; i < DUMB_IT_N_NNA_CHANNELS; i++)
 			if (sigrenderer->playing[i])
-				free(sigrenderer->playing[i]);
+				free_playing_orig(sigrenderer->playing[i]);
 
 		for (playing = sigrenderer->free_playing; playing != NULL; playing = next)
 		{
 			next = playing->next;
-			free(playing);
+			free_playing_orig(playing);
 		}
 
 		dumb_destroy_click_remover_array(sigrenderer->n_channels, sigrenderer->click_remover);
diff --git a/dumb/vc6/dumb_static/dumb_static.vcproj b/dumb/vc6/dumb_static/dumb_static.vcproj
index 128b3b9d9..896ce8a8f 100644
--- a/dumb/vc6/dumb_static/dumb_static.vcproj
+++ b/dumb/vc6/dumb_static/dumb_static.vcproj
@@ -877,6 +877,10 @@
 					RelativePath="..\..\src\helpers\barray.c"
 					>
 				</File>
+				<File
+					RelativePath="..\..\src\helpers\blip_buf.c"
+					>
+				</File>
 				<File
 					RelativePath="..\..\src\helpers\clickrem.c"
 					>
@@ -1960,6 +1964,10 @@
 			<Filter
 				Name="internal"
 				>
+				<File
+					RelativePath="..\..\include\internal\blip_buf.h"
+					>
+				</File>
 				<File
 					RelativePath="..\..\include\internal\dumb.h"
 					>