--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/liboggplayer-src/src/SDL_audiocvt.cpp	Mon Aug 05 13:50:20 2013 +0300
@@ -0,0 +1,1509 @@
+/*
+	This is a modified version of SDL_audiocvt.c from SDL 1.2.13
+	It does not depend on any other files from SDL
+*/
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2006 Sam Lantinga
+
+    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 library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+
+#include "SDL_audiocvt.hpp"
+
+/* Effectively mix right and left channels into a single channel */
+void  SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Sint32 sample;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting to mono\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			for ( i=cvt->len_cvt/2; i; --i ) {
+				sample = src[0] + src[1];
+				*dst = (Uint8)(sample / 2);
+				src += 2;
+				dst += 1;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst;
+
+			src = (Sint8 *)cvt->buf;
+			dst = (Sint8 *)cvt->buf;
+			for ( i=cvt->len_cvt/2; i; --i ) {
+				sample = src[0] + src[1];
+				*dst = (Sint8)(sample / 2);
+				src += 2;
+				dst += 1;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					sample = (Uint16)((src[0]<<8)|src[1])+
+					         (Uint16)((src[2]<<8)|src[3]);
+					sample /= 2;
+					dst[1] = (sample&0xFF);
+					sample >>= 8;
+					dst[0] = (sample&0xFF);
+					src += 4;
+					dst += 2;
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					sample = (Uint16)((src[1]<<8)|src[0])+
+					         (Uint16)((src[3]<<8)|src[2]);
+					sample /= 2;
+					dst[0] = (sample&0xFF);
+					sample >>= 8;
+					dst[1] = (sample&0xFF);
+					src += 4;
+					dst += 2;
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					sample = (Sint16)((src[0]<<8)|src[1])+
+					         (Sint16)((src[2]<<8)|src[3]);
+					sample /= 2;
+					dst[1] = (sample&0xFF);
+					sample >>= 8;
+					dst[0] = (sample&0xFF);
+					src += 4;
+					dst += 2;
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					sample = (Sint16)((src[1]<<8)|src[0])+
+					         (Sint16)((src[3]<<8)|src[2]);
+					sample /= 2;
+					dst[0] = (sample&0xFF);
+					sample >>= 8;
+					dst[1] = (sample&0xFF);
+					src += 4;
+					dst += 2;
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Discard top 4 channels */
+void  SDL_ConvertStrip(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Sint32 lsample, rsample;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting down to stereo\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			for ( i=cvt->len_cvt/6; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 6;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst;
+
+			src = (Sint8 *)cvt->buf;
+			dst = (Sint8 *)cvt->buf;
+			for ( i=cvt->len_cvt/6; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 6;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Uint16)((src[0]<<8)|src[1]);
+					rsample = (Uint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Uint16)((src[1]<<8)|src[0]);
+					rsample = (Uint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Sint16)((src[0]<<8)|src[1]);
+					rsample = (Sint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Sint16)((src[1]<<8)|src[0]);
+					rsample = (Sint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt /= 3;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Discard top 2 channels of 6 */
+void  SDL_ConvertStrip_2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Sint32 lsample, rsample;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting 6 down to quad\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 4;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst;
+
+			src = (Sint8 *)cvt->buf;
+			dst = (Sint8 *)cvt->buf;
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 4;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Uint16)((src[0]<<8)|src[1]);
+					rsample = (Uint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Uint16)((src[1]<<8)|src[0]);
+					rsample = (Uint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Sint16)((src[0]<<8)|src[1]);
+					rsample = (Sint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Sint16)((src[1]<<8)|src[0]);
+					rsample = (Sint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Duplicate a mono channel to both stereo channels */
+void  SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting to stereo\n");
+#endif
+	if ( (format & 0xFF) == 16 ) {
+		Uint16 *src, *dst;
+
+		src = (Uint16 *)(cvt->buf+cvt->len_cvt);
+		dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2);
+		for ( i=cvt->len_cvt/2; i; --i ) {
+			dst -= 2;
+			src -= 1;
+			dst[0] = src[0];
+			dst[1] = src[0];
+		}
+	} else {
+		Uint8 *src, *dst;
+
+		src = cvt->buf+cvt->len_cvt;
+		dst = cvt->buf+cvt->len_cvt*2;
+		for ( i=cvt->len_cvt; i; --i ) {
+			dst -= 2;
+			src -= 1;
+			dst[0] = src[0];
+			dst[1] = src[0];
+		}
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Duplicate a stereo channel to a pseudo-5.1 stream */
+void  SDL_ConvertSurround(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting stereo to surround\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst, lf, rf, ce;
+
+			src = (Uint8 *)(cvt->buf+cvt->len_cvt);
+			dst = (Uint8 *)(cvt->buf+cvt->len_cvt*3);
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 6;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+				dst[4] = ce;
+				dst[5] = ce;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst, lf, rf, ce;
+
+			src = (Sint8 *)cvt->buf+cvt->len_cvt;
+			dst = (Sint8 *)cvt->buf+cvt->len_cvt*3;
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 6;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+				dst[4] = ce;
+				dst[5] = ce;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+			Uint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*3;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Uint16)((src[0]<<8)|src[1]);
+					rf = (Uint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+
+						dst[1+8] = (ce&0xFF);
+						dst[0+8] = ((ce>>8)&0xFF);
+						dst[3+8] = (ce&0xFF);
+						dst[2+8] = ((ce>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Uint16)((src[1]<<8)|src[0]);
+					rf = (Uint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+
+						dst[0+8] = (ce&0xFF);
+						dst[1+8] = ((ce>>8)&0xFF);
+						dst[2+8] = (ce&0xFF);
+						dst[3+8] = ((ce>>8)&0xFF);
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+			Sint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*3;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Sint16)((src[0]<<8)|src[1]);
+					rf = (Sint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+
+						dst[1+8] = (ce&0xFF);
+						dst[0+8] = ((ce>>8)&0xFF);
+						dst[3+8] = (ce&0xFF);
+						dst[2+8] = ((ce>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Sint16)((src[1]<<8)|src[0]);
+					rf = (Sint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+
+						dst[0+8] = (ce&0xFF);
+						dst[1+8] = ((ce>>8)&0xFF);
+						dst[2+8] = (ce&0xFF);
+						dst[3+8] = ((ce>>8)&0xFF);
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt *= 3;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Duplicate a stereo channel to a pseudo-4.0 stream */
+void  SDL_ConvertSurround_4(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting stereo to quad\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst, lf, rf, ce;
+
+			src = (Uint8 *)(cvt->buf+cvt->len_cvt);
+			dst = (Uint8 *)(cvt->buf+cvt->len_cvt*2);
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 4;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst, lf, rf, ce;
+
+			src = (Sint8 *)cvt->buf+cvt->len_cvt;
+			dst = (Sint8 *)cvt->buf+cvt->len_cvt*2;
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 4;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+			Uint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*2;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Uint16)((src[0]<<8)|src[1]);
+					rf = (Uint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Uint16)((src[1]<<8)|src[0]);
+					rf = (Uint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+			Sint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*2;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Sint16)((src[0]<<8)|src[1]);
+					rf = (Sint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Sint16)((src[1]<<8)|src[0]);
+					rf = (Sint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert 8-bit to 16-bit - LSB */
+void  SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting to 16-bit LSB\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	for ( i=cvt->len_cvt; i; --i ) {
+		src -= 1;
+		dst -= 2;
+		dst[1] = *src;
+		dst[0] = 0;
+	}
+	format = ((format & ~0x0008) | AUDIO_U16LSB);
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+/* Convert 8-bit to 16-bit - MSB */
+void  SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting to 16-bit MSB\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	for ( i=cvt->len_cvt; i; --i ) {
+		src -= 1;
+		dst -= 2;
+		dst[0] = *src;
+		dst[1] = 0;
+	}
+	format = ((format & ~0x0008) | AUDIO_U16MSB);
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert 16-bit to 8-bit */
+void  SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting to 8-bit\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
+		++src;
+	}
+	for ( i=cvt->len_cvt/2; i; --i ) {
+		*dst = *src;
+		src += 2;
+		dst += 1;
+	}
+	format = ((format & ~0x9010) | AUDIO_U8);
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Toggle signed/unsigned */
+void  SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *data;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio signedness\n");
+#endif
+	data = cvt->buf;
+	if ( (format & 0xFF) == 16 ) {
+		if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
+			++data;
+		}
+		for ( i=cvt->len_cvt/2; i; --i ) {
+			*data ^= 0x80;
+			data += 2;
+		}
+	} else {
+		for ( i=cvt->len_cvt; i; --i ) {
+			*data++ ^= 0x80;
+		}
+	}
+	format = (format ^ 0x8000);
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Toggle endianness */
+void  SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *data, tmp;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio endianness\n");
+#endif
+	data = cvt->buf;
+	for ( i=cvt->len_cvt/2; i; --i ) {
+		tmp = data[0];
+		data[0] = data[1];
+		data[1] = tmp;
+		data += 2;
+	}
+	format = (format ^ 0x1000);
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert rate up by multiple of 2 */
+void  SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt; i; --i ) {
+				src -= 1;
+				dst -= 2;
+				dst[0] = src[0];
+				dst[1] = src[0];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/2; i; --i ) {
+				src -= 2;
+				dst -= 4;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[0];
+				dst[3] = src[1];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert rate up by multiple of 2, for stereo */
+void  SDL_RateMUL2_c2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/2; i; --i ) {
+				src -= 2;
+				dst -= 4;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[0];
+				dst[3] = src[1];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				src -= 4;
+				dst -= 8;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[0];
+				dst[5] = src[1];
+				dst[6] = src[2];
+				dst[7] = src[3];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert rate up by multiple of 2, for quad */
+void  SDL_RateMUL2_c4(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				src -= 4;
+				dst -= 8;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[0];
+				dst[5] = src[1];
+				dst[6] = src[2];
+				dst[7] = src[3];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/8; i; --i ) {
+				src -= 8;
+				dst -= 16;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				dst[8] = src[0];
+				dst[9] = src[1];
+				dst[10] = src[2];
+				dst[11] = src[3];
+				dst[12] = src[4];
+				dst[13] = src[5];
+				dst[14] = src[6];
+				dst[15] = src[7];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert rate up by multiple of 2, for 5.1 */
+void  SDL_RateMUL2_c6(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/6; i; --i ) {
+				src -= 6;
+				dst -= 12;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[0];
+				dst[7] = src[1];
+				dst[8] = src[2];
+				dst[9] = src[3];
+				dst[10] = src[4];
+				dst[11] = src[5];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/12; i; --i ) {
+				src -= 12;
+				dst -= 24;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				dst[8] = src[8];
+				dst[9] = src[9];
+				dst[10] = src[10];
+				dst[11] = src[11];
+				dst[12] = src[0];
+				dst[13] = src[1];
+				dst[14] = src[2];
+				dst[15] = src[3];
+				dst[16] = src[4];
+				dst[17] = src[5];
+				dst[18] = src[6];
+				dst[19] = src[7];
+				dst[20] = src[8];
+				dst[21] = src[9];
+				dst[22] = src[10];
+				dst[23] = src[11];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert rate down by multiple of 2 */
+void  SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/2; i; --i ) {
+				dst[0] = src[0];
+				src += 2;
+				dst += 1;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 4;
+				dst += 2;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert rate down by multiple of 2, for stereo */
+void  SDL_RateDIV2_c2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 4;
+				dst += 2;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/8; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				src += 8;
+				dst += 4;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert rate down by multiple of 2, for quad */
+void  SDL_RateDIV2_c4(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/8; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				src += 8;
+				dst += 4;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/16; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				src += 16;
+				dst += 8;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert rate down by multiple of 2, for 5.1 */
+void  SDL_RateDIV2_c6(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/12; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				src += 12;
+				dst += 6;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/24; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				dst[8] = src[8];
+				dst[9] = src[9];
+				dst[10] = src[10];
+				dst[11] = src[11];
+				src += 24;
+				dst += 12;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Very slow rate conversion routine */
+void  SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)
+{
+	double ipos;
+	int i, clen;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr);
+#endif
+	clen = (int)((double)cvt->len_cvt / cvt->rate_incr);
+	if ( cvt->rate_incr > 1.0 ) {
+		switch (format & 0xFF) {
+			case 8: {
+				Uint8 *output;
+
+				output = cvt->buf;
+				ipos = 0.0;
+				for ( i=clen; i; --i ) {
+					*output = cvt->buf[(int)ipos];
+					ipos += cvt->rate_incr;
+					output += 1;
+				}
+			}
+			break;
+
+			case 16: {
+				Uint16 *output;
+
+				clen &= ~1;
+				output = (Uint16 *)cvt->buf;
+				ipos = 0.0;
+				for ( i=clen/2; i; --i ) {
+					*output=((Uint16 *)cvt->buf)[(int)ipos];
+					ipos += cvt->rate_incr;
+					output += 1;
+				}
+			}
+			break;
+		}
+	} else {
+		switch (format & 0xFF) {
+			case 8: {
+				Uint8 *output;
+
+				output = cvt->buf+clen;
+				ipos = (double)cvt->len_cvt;
+				for ( i=clen; i; --i ) {
+					ipos -= cvt->rate_incr;
+					output -= 1;
+					*output = cvt->buf[(int)ipos];
+				}
+			}
+			break;
+
+			case 16: {
+				Uint16 *output;
+
+				clen &= ~1;
+				output = (Uint16 *)(cvt->buf+clen);
+				ipos = (double)cvt->len_cvt/2;
+				for ( i=clen/2; i; --i ) {
+					ipos -= cvt->rate_incr;
+					output -= 1;
+					*output=((Uint16 *)cvt->buf)[(int)ipos];
+				}
+			}
+			break;
+		}
+	}
+	cvt->len_cvt = clen;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+int SDL_ConvertAudio(SDL_AudioCVT *cvt)
+{
+	/* Make sure there's data to convert */
+	if ( cvt->buf == NULL ) {
+		return(-1);
+	}
+	/* Return okay if no conversion is necessary */
+	cvt->len_cvt = cvt->len;
+	if ( cvt->filters[0] == NULL ) {
+		return(0);
+	}
+
+	/* Set up the conversion and go! */
+	cvt->filter_index = 0;
+	cvt->filters[0](cvt, cvt->src_format);
+	return(0);
+}
+
+/* Creates a set of audio filters to convert from one format to another.
+   Returns -1 if the format conversion is not supported, or 1 if the
+   audio filter is set up.
+*/
+
+int SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
+	Uint16 src_format, Uint8 src_channels, int src_rate,
+	Uint16 dst_format, Uint8 dst_channels, int dst_rate)
+{
+/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
+		src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/
+	/* Start off with no conversion necessary */
+	cvt->needed = 0;
+	cvt->filter_index = 0;
+	cvt->filters[0] = NULL;
+	cvt->len_mult = 1;
+	cvt->len_ratio = 1.0;
+
+	/* First filter:  Endian conversion from src to dst */
+	if ( (src_format & 0x1000) != (dst_format & 0x1000)
+	     && ((src_format & 0xff) == 16) && ((dst_format & 0xff) == 16)) {
+		cvt->filters[cvt->filter_index++] = SDL_ConvertEndian;
+	}
+
+	/* Second filter: Sign conversion -- signed/unsigned */
+	if ( (src_format & 0x8000) != (dst_format & 0x8000) ) {
+		cvt->filters[cvt->filter_index++] = SDL_ConvertSign;
+	}
+
+	/* Next filter:  Convert 16 bit <--> 8 bit PCM */
+	if ( (src_format & 0xFF) != (dst_format & 0xFF) ) {
+		switch (dst_format&0x10FF) {
+			case AUDIO_U8:
+				cvt->filters[cvt->filter_index++] =
+							 SDL_Convert8;
+				cvt->len_ratio /= 2;
+				break;
+			case AUDIO_U16LSB:
+				cvt->filters[cvt->filter_index++] =
+							SDL_Convert16LSB;
+				cvt->len_mult *= 2;
+				cvt->len_ratio *= 2;
+				break;
+			case AUDIO_U16MSB:
+				cvt->filters[cvt->filter_index++] =
+							SDL_Convert16MSB;
+				cvt->len_mult *= 2;
+				cvt->len_ratio *= 2;
+				break;
+		}
+	}
+
+	/* Last filter:  Mono/Stereo conversion */
+	if ( src_channels != dst_channels ) {
+		if ( (src_channels == 1) && (dst_channels > 1) ) {
+			cvt->filters[cvt->filter_index++] =
+						SDL_ConvertStereo;
+			cvt->len_mult *= 2;
+			src_channels = 2;
+			cvt->len_ratio *= 2;
+		}
+		if ( (src_channels == 2) &&
+				(dst_channels == 6) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertSurround;
+			src_channels = 6;
+			cvt->len_mult *= 3;
+			cvt->len_ratio *= 3;
+		}
+		if ( (src_channels == 2) &&
+				(dst_channels == 4) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertSurround_4;
+			src_channels = 4;
+			cvt->len_mult *= 2;
+			cvt->len_ratio *= 2;
+		}
+		while ( (src_channels*2) <= dst_channels ) {
+			cvt->filters[cvt->filter_index++] =
+						SDL_ConvertStereo;
+			cvt->len_mult *= 2;
+			src_channels *= 2;
+			cvt->len_ratio *= 2;
+		}
+		if ( (src_channels == 6) &&
+				(dst_channels <= 2) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertStrip;
+			src_channels = 2;
+			cvt->len_ratio /= 3;
+		}
+		if ( (src_channels == 6) &&
+				(dst_channels == 4) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertStrip_2;
+			src_channels = 4;
+			cvt->len_ratio /= 2;
+		}
+		/* This assumes that 4 channel audio is in the format:
+		     Left {front/back} + Right {front/back}
+		   so converting to L/R stereo works properly.
+		 */
+		while ( ((src_channels%2) == 0) &&
+				((src_channels/2) >= dst_channels) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertMono;
+			src_channels /= 2;
+			cvt->len_ratio /= 2;
+		}
+		if ( src_channels != dst_channels ) {
+			/* Uh oh.. */;
+		}
+	}
+
+	/* Do rate conversion */
+	cvt->rate_incr = 0.0;
+	if ( (src_rate/100) != (dst_rate/100) ) {
+		Uint32 hi_rate, lo_rate;
+		int len_mult;
+		double len_ratio;
+		void ( *rate_cvt)(SDL_AudioCVT *cvt, Uint16 format);
+
+		if ( src_rate > dst_rate ) {
+			hi_rate = src_rate;
+			lo_rate = dst_rate;
+			switch (src_channels) {
+				case 1: rate_cvt = SDL_RateDIV2; break;
+				case 2: rate_cvt = SDL_RateDIV2_c2; break;
+				case 4: rate_cvt = SDL_RateDIV2_c4; break;
+				case 6: rate_cvt = SDL_RateDIV2_c6; break;
+				default: return -1;
+			}
+			len_mult = 1;
+			len_ratio = 0.5;
+		} else {
+			hi_rate = dst_rate;
+			lo_rate = src_rate;
+			switch (src_channels) {
+				case 1: rate_cvt = SDL_RateMUL2; break;
+				case 2: rate_cvt = SDL_RateMUL2_c2; break;
+				case 4: rate_cvt = SDL_RateMUL2_c4; break;
+				case 6: rate_cvt = SDL_RateMUL2_c6; break;
+				default: return -1;
+			}
+			len_mult = 2;
+			len_ratio = 2.0;
+		}
+		/* If hi_rate = lo_rate*2^x then conversion is easy */
+		while ( ((lo_rate*2)/100) <= (hi_rate/100) ) {
+			cvt->filters[cvt->filter_index++] = rate_cvt;
+			cvt->len_mult *= len_mult;
+			lo_rate *= 2;
+			cvt->len_ratio *= len_ratio;
+		}
+		/* We may need a slow conversion here to finish up */
+		if ( (lo_rate/100) != (hi_rate/100) ) {
+#if 1
+			/* The problem with this is that if the input buffer is
+			   say 1K, and the conversion rate is say 1.1, then the
+			   output buffer is 1.1K, which may not be an acceptable
+			   buffer size for the audio driver (not a power of 2)
+			*/
+			/* For now, punt and hope the rate distortion isn't great.
+			*/
+#else
+			if ( src_rate < dst_rate ) {
+				cvt->rate_incr = (double)lo_rate/hi_rate;
+				cvt->len_mult *= 2;
+				cvt->len_ratio /= cvt->rate_incr;
+			} else {
+				cvt->rate_incr = (double)hi_rate/lo_rate;
+				cvt->len_ratio *= cvt->rate_incr;
+			}
+			cvt->filters[cvt->filter_index++] = SDL_RateSLOW;
+#endif
+		}
+	}
+
+	/* Set up the filter information */
+	if ( cvt->filter_index != 0 ) {
+		cvt->needed = 1;
+		cvt->src_format = src_format;
+		cvt->dst_format = dst_format;
+		cvt->len = 0;
+		cvt->buf = NULL;
+		cvt->filters[cvt->filter_index] = NULL;
+	}
+	return(cvt->needed);
+}