Mercurial > sdl-ios-xcode
view src/audio/SDL_mixer.c @ 641:df178851293b
Date: 28 Jun 2003 22:42:52 +0100
From: Alan Swanson
Subject: Re: [SDL] New XFree 4.3 Video Mode Patch
I have a wee amendment that moves the qsort in set_best_resolution
to only occur after failing to find an exact match only. This would
make absolutely sure we get a user set mode.
While I've never had any problems for my normal resolutions (1280x1024,
1024x768, 800x600 & 640,480) while closely examining the output from
qsort I've noticed it doesn't seem to sort the modes fully. These is
one definite wrong at 1152x768 and a few that just look wrong to me.
From a program (attached) I made to examine this more easily. X has
sorted its mode list using the same method as ours (plus frequency),
and our user modes get inserted without any other movement.
On the patch I've made I've also changed cmpmodes to sort on vertical
resolution and then horizontal. Ie vertical is now most significant
bit.
author | Sam Lantinga <slouken@libsdl.org> |
---|---|
date | Sat, 28 Jun 2003 21:52:26 +0000 |
parents | 873c2598f969 |
children | 22dbf364c017 |
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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_sysaudio.h" #include "SDL_mixer_MMX.h" #include "SDL_mixer_MMX_VC.h" #include "SDL_mixer_m68k.h" /* Function to check the CPU flags */ #define MMX_CPU 0x800000 #ifdef USE_ASMBLIT #define CPU_Flags() Hermes_X86_CPU() #else #define CPU_Flags() 0L #endif #ifdef USE_ASMBLIT #define X86_ASSEMBLER #define HermesConverterInterface void #define HermesClearInterface void #define STACKCALL #include "HeadX86.h" #endif /* 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 (CPU_Flags() & MMX_CPU) { SDL_MixAudio_MMX_S8((char*)dst,(char*)src,(unsigned int)len,(int)volume); } else #endif #if defined(USE_ASM_MIXER_VC) if (SDL_IsMMX_VC()) { 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 (CPU_Flags() & MMX_CPU) { SDL_MixAudio_MMX_S16((char*)dst,(char*)src,(unsigned int)len,(int)volume); } else #elif defined(USE_ASM_MIXER_VC) if (SDL_IsMMX_VC()) { 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; } }