Mercurial > sdl-ios-xcode

view src/audio/SDL_audiocvt.c @ 1192:54aa9aa32327

Find changesets by keywords (author, files, the commit message), revision number or hash, or revset expression.
To: sdl@libsdl.org From: Christian Walther <cwalther@gmx.ch> Date: Fri, 18 Nov 2005 23:39:02 +0100 Subject: [SDL] Mouse position bugs on Mac OS X The attached patch fixes a few bugs in SDL related to the mouse position in windowed mode on Mac OS X, reproduced using the attached minimal test program - at least here on 10.3.9, with SDL CVS from today. Could anyone test whether the bugs exist and are fixed by the patch on 10.2 and 10.4? 1. When using OpenGL, the vertical mouse positions obtained through events or SDL_GetMouseState() are off by one. 2. When using OpenGL, SDL_WarpMouse() inverts the y coordinate. 3. Clicks on the topmost pixel row of the window are not recognized. 1 and 2 do not occur in non-OpenGL mode, while 3 does. All three only occur in windowed mode, not in fullscreen. The cause for 1 and 3 is that in Cocoa, "the location of the mouse" seems to be defined as "the location of the top left corner of the mouse pointer's hot pixel" (this is not documented, it's just what I found out here), which together with the fact that Cocoa's usual y coordinates start at the bottom and increase upwards means that the y coordinate of the mouse runs from 1 to h, not from 0 to h-1, in a window of height h. If it does work on 10.2 and 10.4 (I'll try to test it as soon as I can, but at the moment all I have at hand is 10.3.9), can this be applied to the CVS? -Christian To: sdl@libsdl.org From: Christian Walther <cwalther@gmx.ch> Date: Mon, 28 Nov 2005 10:41:51 +0100 Subject: [SDL] Re: Mouse position bugs on Mac OS X I wrote: > I'll try to test it as soon as I can, but at the moment all I have at hand is 10.3.9 So, here are the results of my tests (with patched and unpatched frameworks compiled with Xcode 1.5 (gcc 3.3) on 10.3.9): On 10.1.5, my test program doesn't run because of "Undefined symbols: SDL undefined reference to _CGMainDisplayID expected to be defined in Carbon". I guess not supporting 10.1 was a deliberate decision then and that's OK with me. On 10.2.8, 10.3.9, and 10.4.0, the bugs exist as described in my original post and are fixed by my patch. That is, there is no difference between pre/post 10.3 and the patched version works correctly in all combinations of GL/non-GL and windowed/fullscreen. I therefore recommend the patch for inclusion. -Christian
author Ryan C. Gordon <icculus@icculus.org>
date Mon, 28 Nov 2005 13:58:26 +0000
parents 4095d9ca23f2
children c9b51268668f
line wrap: on
line source

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2004 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Library General Public
    License as published by the Free Software Foundation; either
    version 2 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
    Library General Public License for more details.

    You should have received a copy of the GNU Library General Public
    License along with this library; if not, write to the Free
    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

    Sam Lantinga
    slouken@libsdl.org
*/

#ifdef SAVE_RCSID
static char rcsid =
 "@(#) $Id$";
#endif

/* Functions for audio drivers to perform runtime conversion of audio format */

#include <stdio.h>

#include "SDL_error.h"
#include "SDL_audio.h"


/* 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];
				if ( sample > 255 ) {
					*dst = 255;
				} else {
					*dst = sample;
				}
				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];
				if ( sample > 127 ) {
					*dst = 127;
				} else
				if ( sample < -128 ) {
					*dst = -128;
				} else {
					*dst = sample;
				}
				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]);
					if ( sample > 65535 ) {
						dst[0] = 0xFF;
						dst[1] = 0xFF;
					} else {
						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]);
					if ( sample > 65535 ) {
						dst[0] = 0xFF;
						dst[1] = 0xFF;
					} else {
						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]);
					if ( sample > 32767 ) {
						dst[0] = 0x7F;
						dst[1] = 0xFF;
					} else
					if ( sample < -32768 ) {
						dst[0] = 0x80;
						dst[1] = 0x00;
					} else {
						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]);
					if ( sample > 32767 ) {
						dst[1] = 0x7F;
						dst[0] = 0xFF;
					} else
					if ( sample < -32768 ) {
						dst[1] = 0x80;
						dst[0] = 0x00;
					} else {
						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 ) {
				lsample = src[0];
				rsample = src[1];
				dst[0] = lsample;
				dst[1] = rsample;
				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 ) {
				lsample = src[0];
				rsample = src[1];
				dst[0] = lsample;
				dst[1] = rsample;
				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 ) {
				lsample = src[0];
				rsample = src[1];
				dst[0] = lsample;
				dst[1] = rsample;
				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 ) {
				lsample = src[0];
				rsample = src[1];
				dst[0] = lsample;
				dst[1] = rsample;
				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 ) {
		SDL_SetError("No buffer allocated for conversion");
		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) != 8) ) {
		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);
}