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Date: Sun, 4 Jan 2004 23:48:19 +0100 From: Max Horn Subject: Re: Again Audio CD patch Am 04.01.2004 um 22:38 schrieb Sam Lantinga: > > Okay, I fixed the buffering problems by simply using a 4 second buffer > instead of a 1 second buffer. However, using your code I can't play an > entire CD - the playback stops after the first song. > Found the problem: FSReadFork returns eofErr when the file is finished. However, we check its return value for errors, and if anything but noErr occurs, the reader thread aborts its current iteration. That is bad, because it aborts before it can ever set the flag which tells that the file is over (also, any remaining data which FSRead did return is lost - so you'd not hear about to 4 seconds from the end of the file. Furthermore, the computed data size was 8 bytes to high (I forgot to account for the fact that the size of an (A)IFF chunk always contains the chunk header & size fields, too). This is enough to make it work. However, the end condition is rather fragile, so I tuned some other things to be pessimistic (check for <= 0 instead of == 0, when eofErr is encountered enforce mReadFilePosition == mFileLength). You never know... The attached patch fixes the issue for me.
author Sam Lantinga <slouken@libsdl.org>
date Mon, 05 Jan 2004 00:57:51 +0000
parents b8d311d90021
children 41a59de7f2ed
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);
	}
}


/* 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);
	}
}

/* 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 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);
	}
}

/* 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)
{
	/* 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 ) {
		while ( (src_channels*2) <= dst_channels ) {
			cvt->filters[cvt->filter_index++] = 
						SDL_ConvertStereo;
			cvt->len_mult *= 2;
			src_channels *= 2;
			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;
			rate_cvt = SDL_RateDIV2;
			len_mult = 1;
			len_ratio = 0.5;
		} else {
			hi_rate = dst_rate;
			lo_rate = src_rate;
			rate_cvt = SDL_RateMUL2;
			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);
}