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Date: Mon, 30 Aug 2004 18:20:25 +0200 From: Joost Baas Subject: why call arts artsc? I, and a few other people at the mplayer-docs-mailinglist were wondering why you decided to call arts artsc. I understand usually users have nothing to do with libsdl, just developers, but because you can choose the audio-driver being used by mplayer, one of which is sdl, and you can also choose the sdl subdriver, it is necessary to have a well-known or logical name. artsc is not the logical choice, and it's very hard to look up the right name if you don't know what you're looking for.
author Sam Lantinga <slouken@libsdl.org>
date Fri, 17 Sep 2004 13:25:06 +0000
parents b8d311d90021
children c9b51268668f
line wrap: on
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/*
    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

/* This provides the default mixing callback for the SDL audio routines */

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#include "SDL_audio.h"
#include "SDL_mutex.h"
#include "SDL_timer.h"
#include "SDL_cpuinfo.h"
#include "SDL_sysaudio.h"
#include "SDL_cpuinfo.h"
#include "SDL_mixer_MMX.h"
#include "SDL_mixer_MMX_VC.h"
#include "SDL_mixer_m68k.h"

/* This table is used to add two sound values together and pin
 * the value to avoid overflow.  (used with permission from ARDI)
 * Changed to use 0xFE instead of 0xFF for better sound quality.
 */
static const Uint8 mix8[] =
{
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
  0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
  0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
  0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
  0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
  0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
  0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
  0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
  0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
  0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
  0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
  0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
  0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
  0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
  0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
  0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
  0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
  0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
  0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
  0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
  0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
  0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
  0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
  0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE
};

/* The volume ranges from 0 - 128 */
#define ADJUST_VOLUME(s, v)	(s = (s*v)/SDL_MIX_MAXVOLUME)
#define ADJUST_VOLUME_U8(s, v)	(s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)

void SDL_MixAudio (Uint8 *dst, const Uint8 *src, Uint32 len, int volume)
{
	Uint16 format;

	if ( volume == 0 ) {
		return;
	}
	/* Mix the user-level audio format */
	if ( current_audio ) {
		if ( current_audio->convert.needed ) {
			format = current_audio->convert.src_format;
		} else {
			format = current_audio->spec.format;
		}
	} else {
  		/* HACK HACK HACK */
		format = AUDIO_S16;
	}
	switch (format) {

		case AUDIO_U8: {
#if defined(__M68000__) && defined(__GNUC__)
			SDL_MixAudio_m68k_U8((char*)dst,(char*)src,(unsigned long)len,(long)volume,(char *)mix8);
#else
			Uint8 src_sample;

			while ( len-- ) {
				src_sample = *src;
				ADJUST_VOLUME_U8(src_sample, volume);
				*dst = mix8[*dst+src_sample];
				++dst;
				++src;
			}
#endif
		}
		break;

		case AUDIO_S8: {
#if defined(i386) && defined(__GNUC__) && defined(USE_ASMBLIT)
			if (SDL_HasMMX())
			{
				SDL_MixAudio_MMX_S8((char*)dst,(char*)src,(unsigned int)len,(int)volume);
			}
			else
#endif
#if defined(USE_ASM_MIXER_VC)
			if (SDL_HasMMX())
			{
				SDL_MixAudio_MMX_S8_VC((char*)dst,(char*)src,(unsigned int)len,(int)volume);
			}
			else
#endif
#if defined(__M68000__) && defined(__GNUC__)
			SDL_MixAudio_m68k_S8((char*)dst,(char*)src,(unsigned long)len,(long)volume);
#else
			{
			Sint8 *dst8, *src8;
			Sint8 src_sample;
			int dst_sample;
			const int max_audioval = ((1<<(8-1))-1);
			const int min_audioval = -(1<<(8-1));

			src8 = (Sint8 *)src;
			dst8 = (Sint8 *)dst;
			while ( len-- ) {
				src_sample = *src8;
				ADJUST_VOLUME(src_sample, volume);
				dst_sample = *dst8 + src_sample;
				if ( dst_sample > max_audioval ) {
					*dst8 = max_audioval;
				} else
				if ( dst_sample < min_audioval ) {
					*dst8 = min_audioval;
				} else {
					*dst8 = dst_sample;
				}
				++dst8;
				++src8;
			}
			}
#endif
		}
		break;

		case AUDIO_S16LSB: {
#if defined(i386) && defined(__GNUC__) && defined(USE_ASMBLIT)
			if (SDL_HasMMX())
			{
				SDL_MixAudio_MMX_S16((char*)dst,(char*)src,(unsigned int)len,(int)volume);
			}
			else
#elif defined(USE_ASM_MIXER_VC)
			if (SDL_HasMMX())
			{
				SDL_MixAudio_MMX_S16_VC((char*)dst,(char*)src,(unsigned int)len,(int)volume);
			}
			else
#endif
#if defined(__M68000__) && defined(__GNUC__)
			SDL_MixAudio_m68k_S16LSB((short*)dst,(short*)src,(unsigned long)len,(long)volume);
#else
			{
			Sint16 src1, src2;
			int dst_sample;
			const int max_audioval = ((1<<(16-1))-1);
			const int min_audioval = -(1<<(16-1));

			len /= 2;
			while ( len-- ) {
				src1 = ((src[1])<<8|src[0]);
				ADJUST_VOLUME(src1, volume);
				src2 = ((dst[1])<<8|dst[0]);
				src += 2;
				dst_sample = src1+src2;
				if ( dst_sample > max_audioval ) {
					dst_sample = max_audioval;
				} else
				if ( dst_sample < min_audioval ) {
					dst_sample = min_audioval;
				}
				dst[0] = dst_sample&0xFF;
				dst_sample >>= 8;
				dst[1] = dst_sample&0xFF;
				dst += 2;
			}
			}
#endif
		}
		break;

		case AUDIO_S16MSB: {
#if defined(__M68000__) && defined(__GNUC__)
			SDL_MixAudio_m68k_S16MSB((short*)dst,(short*)src,(unsigned long)len,(long)volume);
#else
			Sint16 src1, src2;
			int dst_sample;
			const int max_audioval = ((1<<(16-1))-1);
			const int min_audioval = -(1<<(16-1));

			len /= 2;
			while ( len-- ) {
				src1 = ((src[0])<<8|src[1]);
				ADJUST_VOLUME(src1, volume);
				src2 = ((dst[0])<<8|dst[1]);
				src += 2;
				dst_sample = src1+src2;
				if ( dst_sample > max_audioval ) {
					dst_sample = max_audioval;
				} else
				if ( dst_sample < min_audioval ) {
					dst_sample = min_audioval;
				}
				dst[1] = dst_sample&0xFF;
				dst_sample >>= 8;
				dst[0] = dst_sample&0xFF;
				dst += 2;
			}
#endif
		}
		break;

		default: /* If this happens... FIXME! */
			SDL_SetError("SDL_MixAudio(): unknown audio format");
			return;
	}
}