sdl_audiocvt.c

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}


/* 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 Our_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);
}