Mercurial > sdl-ios-xcode
view src/audio/SDL_mixer.c @ 1118:65f4b2dd46b7
Date: Wed, 17 Aug 2005 11:23:40 -0400
From: Matt L <prometheus.uw@gmail.com>
To: "Ryan C. Gordon" <icculus@clutteredmind.org>
Subject: SDL Patch
Hio,
I saw your last call on the mailing list. Here's a patch which I
submitted about two weeks ago which hasn't made it in.
In the current sdl.m4, there's a bug where if your configure.ac, you
have AC_LANG(C++) specified, it won't properly compile and link the SDL
test program when you run the configure script. This is because only the
default CFLAGS is overriden in sdl.m4, and in the patch below, I've
fixed it so it overrides CXXFLAGS as well, allowing it to work with g++.
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Thu, 18 Aug 2005 06:06:02 +0000 |
parents | b8d311d90021 |
children | c9b51268668f |
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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; } }