view src/audio/SDL_mixer.c @ 753:b14fdadd8311

Date: Thu, 4 Dec 2003 07:48:40 +0200 From: "Mike Gorchak" Subject: SDL/QNX6 new patch Here in attachment my patch for the SDL/QNX6 again :) It contain non-crtitical/cosmetic fixes: 1. Fixed window centering at other than the first consoles. 2. Fixed window centering algorithm in case when window height or width are greater than the desktop resolution. 3. Fixed window positioning on other than the first consoles. 4. Fixed occasional input focus lost when switching to fullscreen. 5. Removed the Photon's default chroma color for the overlays, added RGB(12, 6, 12) color instead (very dark pink). 6. Added more checks to the YUV overlay code (fixed crashes during resolution mode switches). 7. Added support for Enter/Backspace keys in unicode mode (used by Maelstrom and by other games). 8. Fixed window restore/maximize function. It works, finally.
author Sam Lantinga <slouken@libsdl.org>
date Wed, 10 Dec 2003 12:35:56 +0000
parents 72ef7ce609ef
children b8d311d90021
line wrap: on
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/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997, 1998, 1999, 2000, 2001, 2002  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;
	}
}