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view src/audio/SDL_audiocvt.c @ 4324:1496aa09e41e SDL-1.2

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Steven Noonan to sdl While trying to build the SDLMain.m included with SDL 1.2.14, with #define SDL_USE_NIB_FILE 1: /Users/steven/Development/darwinia/targets/macosx/Darwinia/SDLMain.m: In function '-[SDLMain fixMenu:withAppName:]': /Users/steven/Development/darwinia/targets/macosx/Darwinia/SDLMain.m:122: warning: 'sizeToFit' is deprecated (declared at /Developer/SDKs/MacOSX10.6.sdk/System/Library/Frameworks/AppKit.framework/Headers/NSMenu.h:281) /Users/steven/Development/darwinia/targets/macosx/Darwinia/SDLMain.m: In function 'main': /Users/steven/Development/darwinia/targets/macosx/Darwinia/SDLMain.m:376: warning: 'poseAsClass:' is deprecated (declared at /Developer/SDKs/MacOSX10.6.sdk/System/Library/Frameworks/Foundation.framework/Headers/NSObject.h:127) /Users/steven/Development/darwinia/targets/macosx/Darwinia/SDLMain.m:376: error: 'poseAsClass:' is unavailable (declared at /Developer/SDKs/MacOSX10.6.sdk/System/Library/Frameworks/Foundation.framework/Headers/NSObject.h:127) /Users/steven/Development/darwinia/targets/macosx/Darwinia/SDLMain.m:377: warning: passing argument 2 of 'NSApplicationMain' from incompatible pointer type Eric Wing to Sam I don't have time today to look at this in detail, but the problem is definitely the poseAsClass: method. This was deprecated in Obj-C 2.0 and not retained in 64-bit. I've never used this method and it has always been limited to esoteric uses. I think this is why Apple wanted to dump it (among complicating some other things they do). I have read about others getting bit by this when migrating. Long story short, there really isn't a migration path for this method. The question that then must be asked is why are we using it (what does it accomplish), and then figure out the 'proper' way of accomplishing that. Glancing at SDLMain.m, it's not obvious to me why it is there or what it is really accomplishing. My only speculation is that NSApplicationMain hardcodes something to look for NSApplication and a subclass (SDLApplication) fails for some reason (assuming that the original coder did this for good reason). Three thoughts come to mind. 1) The Info.plist has properties to control things related to the startup class and nib. NSPrincipalClass, NSMainNibFile Maybe principle class needs to be SDLApplication and we can delete the poseAs 2) I was told that 10.6 introduced new APIs to make programatic NIB wrangling and avoidance easier. Unfortunately, I don't know the specifics. 3) Instead of subclassing NSApplication in SDLMain.m, maybe we can just add a category. It looks like the following is the only thing that is done (quick glance): @interface SDLApplication : NSApplication @end @implementation SDLApplication /* Invoked from the Quit menu item */ - (void)terminate:(id)sender { /* Post a SDL_QUIT event */ SDL_Event event; event.type = SDL_QUIT; SDL_PushEvent(&event); } @end So instead, we change this to: (warning written in mail and untested) @interface NSApplication (SDLApplication) - (void) terminate:(id)sender; @end @implementation NSApplication (SDLApplication) /* Invoked from the Quit menu item */ - (void)terminate:(id)sender { /* Post a SDL_QUIT event */ SDL_Event event; event.type = SDL_QUIT; SDL_PushEvent(&event); } @end Then everywhere SDLApplication is used, we change it to NSApplication (and remove the poseAsClass line). Perhaps you could ask the bug reporter to try this solution (#3). And if that fails, maybe try #1. -Eric Steven Noonan to Sam The suggested change (diff below) seems to work fine. - Steven
author Sam Lantinga <slouken@libsdl.org>
date Mon, 12 Oct 2009 21:07:12 +0000
parents a1b03ba2fcd0
children
line wrap: on
line source

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 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_config.h"

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

#include "SDL_audio.h"


/* Effectively mix right and left channels into a single channel */
void SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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 SDLCALL 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) == 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 (SDLCALL *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);
}