Mercurial > sdl-ios-xcode
view src/audio/SDL_mixer.c @ 968:4675910b0b7b
Date: Mon, 11 Oct 2004 15:17:27 +0300 (EEST)
From: Hannu Savolainen
Subject: Re: SDL uses obsolete OSS features
I did some work on getting OSS to work better with SDL. There have been
some problems with select which should be fixed now.
I'm having some problems in understanding what is the purpose of the
DSP_WaitAudio() routine. I added a return to the very beginning of this
routine and commendted out the define for USE_BLOCKING_WRITES. At least
lbreakout2 seems to work as well as earlier. The latencies are the same.
An ordinary blocking write does exactly the same thing than DSP_WaitAudio
does. So I would recommend using the USE_BLOCKING_WRITES approach and
removing everything from the DSP_WaitAudio routine. Also enabling
USE_BLOCKING_WRITES makes it possible to simplify DSP_PlayAudio() because
you don't need to handle the partial writes (the do-while loop).
Attached is a patch against SDL-1.2.7. After these changes SDL will use
OSS as it's designed to be used (make it as simple as possible). This code
should work with all OSS implementations because it uses only the very
fundamental features that have been there since the jurassic times.
author | Sam Lantinga <slouken@libsdl.org> |
---|---|
date | Fri, 12 Nov 2004 21:39:04 +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; } }