src/audio/SDL_mixer.c
author Sam Lantinga <slouken@libsdl.org>
Fri, 08 Apr 2011 13:16:33 -0700
changeset 5536 05af1b9ff46d
parent 5535 96594ac5fd1a
child 6138 4c64952a58fb
permissions -rw-r--r--
Removed buggy and LGPL MMX mixing routines.
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/*
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  Simple DirectMedia Layer
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  Copyright (C) 1997-2011 Sam Lantinga <slouken@libsdl.org>
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  This software is provided 'as-is', without any express or implied
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  warranty.  In no event will the authors be held liable for any damages
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  arising from the use of this software.
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  Permission is granted to anyone to use this software for any purpose,
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  including commercial applications, and to alter it and redistribute it
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  freely, subject to the following restrictions:
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  1. The origin of this software must not be misrepresented; you must not
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     claim that you wrote the original software. If you use this software
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     in a product, an acknowledgment in the product documentation would be
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     appreciated but is not required.
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  2. Altered source versions must be plainly marked as such, and must not be
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     misrepresented as being the original software.
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  3. This notice may not be removed or altered from any source distribution.
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*/
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#include "SDL_config.h"
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/* This provides the default mixing callback for the SDL audio routines */
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#include "SDL_cpuinfo.h"
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#include "SDL_timer.h"
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#include "SDL_audio.h"
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#include "SDL_sysaudio.h"
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/* This table is used to add two sound values together and pin
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 * the value to avoid overflow.  (used with permission from ARDI)
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 * Changed to use 0xFE instead of 0xFF for better sound quality.
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 */
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static const Uint8 mix8[] = {
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
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    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
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    0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
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    0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
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    0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
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    0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
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    0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
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    0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
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    0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
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    0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
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    0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
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    0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
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    0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
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    0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
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    0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
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    0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
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    0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
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    0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
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    0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
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    0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
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    0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
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    0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
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    0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
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    0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
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    0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
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    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE
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};
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/* The volume ranges from 0 - 128 */
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#define ADJUST_VOLUME(s, v)	(s = (s*v)/SDL_MIX_MAXVOLUME)
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#define ADJUST_VOLUME_U8(s, v)	(s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)
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void
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SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
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                   Uint32 len, int volume)
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{
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    if (volume == 0) {
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        return;
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    }
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    switch (format) {
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    case AUDIO_U8:
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        {
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#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
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            SDL_MixAudio_m68k_U8((char *) dst, (char *) src,
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                                 (unsigned long) len, (long) volume,
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                                 (char *) mix8);
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#else
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            Uint8 src_sample;
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            while (len--) {
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                src_sample = *src;
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                ADJUST_VOLUME_U8(src_sample, volume);
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                *dst = mix8[*dst + src_sample];
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                ++dst;
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                ++src;
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            }
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#endif
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        }
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        break;
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    case AUDIO_S8:
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        {
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            Sint8 *dst8, *src8;
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            Sint8 src_sample;
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            int dst_sample;
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            const int max_audioval = ((1 << (8 - 1)) - 1);
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            const int min_audioval = -(1 << (8 - 1));
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            src8 = (Sint8 *) src;
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            dst8 = (Sint8 *) dst;
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            while (len--) {
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                src_sample = *src8;
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                ADJUST_VOLUME(src_sample, volume);
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                dst_sample = *dst8 + src_sample;
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                if (dst_sample > max_audioval) {
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                    *dst8 = max_audioval;
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                } else if (dst_sample < min_audioval) {
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                    *dst8 = min_audioval;
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                } else {
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                    *dst8 = dst_sample;
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                }
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                ++dst8;
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                ++src8;
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            }
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        }
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        break;
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    case AUDIO_S16LSB:
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        {
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            Sint16 src1, src2;
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            int dst_sample;
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            const int max_audioval = ((1 << (16 - 1)) - 1);
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            const int min_audioval = -(1 << (16 - 1));
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            len /= 2;
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            while (len--) {
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                src1 = ((src[1]) << 8 | src[0]);
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                ADJUST_VOLUME(src1, volume);
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                src2 = ((dst[1]) << 8 | dst[0]);
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                src += 2;
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                dst_sample = src1 + src2;
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                if (dst_sample > max_audioval) {
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                    dst_sample = max_audioval;
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                } else if (dst_sample < min_audioval) {
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                    dst_sample = min_audioval;
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                }
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                dst[0] = dst_sample & 0xFF;
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                dst_sample >>= 8;
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                dst[1] = dst_sample & 0xFF;
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                dst += 2;
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            }
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        }
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        break;
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    case AUDIO_S16MSB:
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        {
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#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
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            SDL_MixAudio_m68k_S16MSB((short *) dst, (short *) src,
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                                     (unsigned long) len, (long) volume);
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#else
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            Sint16 src1, src2;
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            int dst_sample;
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            const int max_audioval = ((1 << (16 - 1)) - 1);
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            const int min_audioval = -(1 << (16 - 1));
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            len /= 2;
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            while (len--) {
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                src1 = ((src[0]) << 8 | src[1]);
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                ADJUST_VOLUME(src1, volume);
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                src2 = ((dst[0]) << 8 | dst[1]);
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                src += 2;
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                dst_sample = src1 + src2;
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                if (dst_sample > max_audioval) {
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                    dst_sample = max_audioval;
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                } else if (dst_sample < min_audioval) {
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                    dst_sample = min_audioval;
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                }
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                dst[1] = dst_sample & 0xFF;
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                dst_sample >>= 8;
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                dst[0] = dst_sample & 0xFF;
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                dst += 2;
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            }
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#endif
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        }
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        break;
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    case AUDIO_S32LSB:
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        {
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            const Uint32 *src32 = (Uint32 *) src;
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            Uint32 *dst32 = (Uint32 *) dst;
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            Sint64 src1, src2;
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            Sint64 dst_sample;
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            const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
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            const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
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            len /= 4;
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            while (len--) {
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                src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
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                src32++;
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                ADJUST_VOLUME(src1, volume);
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                src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
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                dst_sample = src1 + src2;
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                if (dst_sample > max_audioval) {
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                    dst_sample = max_audioval;
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                } else if (dst_sample < min_audioval) {
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                    dst_sample = min_audioval;
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                }
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                *(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
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            }
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        }
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        break;
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   231
    case AUDIO_S32MSB:
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        {
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            const Uint32 *src32 = (Uint32 *) src;
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            Uint32 *dst32 = (Uint32 *) dst;
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   235
            Sint64 src1, src2;
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   236
            Sint64 dst_sample;
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   237
            const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
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   238
            const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
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   239
icculus@1982
   240
            len /= 4;
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   241
            while (len--) {
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   242
                src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
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                src32++;
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   244
                ADJUST_VOLUME(src1, volume);
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   245
                src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
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   246
                dst_sample = src1 + src2;
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   247
                if (dst_sample > max_audioval) {
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   248
                    dst_sample = max_audioval;
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   249
                } else if (dst_sample < min_audioval) {
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   250
                    dst_sample = min_audioval;
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   251
                }
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   252
                *(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
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   253
            }
icculus@1982
   254
        }
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   255
        break;
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   256
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   257
    case AUDIO_F32LSB:
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   258
        {
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   259
            const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
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   260
            const float fvolume = (float) volume;
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   261
            const float *src32 = (float *) src;
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   262
            float *dst32 = (float *) dst;
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   263
            float src1, src2;
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   264
            double dst_sample;
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   265
            /* !!! FIXME: are these right? */
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            const double max_audioval = 3.402823466e+38F;
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   267
            const double min_audioval = -3.402823466e+38F;
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   268
icculus@1982
   269
            len /= 4;
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   270
            while (len--) {
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   271
                src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
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   272
                src2 = SDL_SwapFloatLE(*dst32);
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   273
                src32++;
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   274
icculus@2014
   275
                dst_sample = ((double) src1) + ((double) src2);
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   276
                if (dst_sample > max_audioval) {
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                    dst_sample = max_audioval;
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   278
                } else if (dst_sample < min_audioval) {
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                    dst_sample = min_audioval;
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                }
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   281
                *(dst32++) = SDL_SwapFloatLE((float) dst_sample);
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   282
            }
icculus@1982
   283
        }
icculus@1982
   284
        break;
icculus@1982
   285
icculus@1982
   286
    case AUDIO_F32MSB:
icculus@1982
   287
        {
icculus@1982
   288
            const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
icculus@1982
   289
            const float fvolume = (float) volume;
icculus@1982
   290
            const float *src32 = (float *) src;
icculus@1982
   291
            float *dst32 = (float *) dst;
icculus@1982
   292
            float src1, src2;
icculus@1982
   293
            double dst_sample;
icculus@1982
   294
            /* !!! FIXME: are these right? */
slouken@2130
   295
            const double max_audioval = 3.402823466e+38F;
slouken@2130
   296
            const double min_audioval = -3.402823466e+38F;
icculus@1982
   297
icculus@1982
   298
            len /= 4;
icculus@1982
   299
            while (len--) {
icculus@2014
   300
                src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
icculus@2014
   301
                src2 = SDL_SwapFloatBE(*dst32);
icculus@2014
   302
                src32++;
icculus@1982
   303
icculus@2014
   304
                dst_sample = ((double) src1) + ((double) src2);
icculus@1982
   305
                if (dst_sample > max_audioval) {
icculus@1982
   306
                    dst_sample = max_audioval;
icculus@1982
   307
                } else if (dst_sample < min_audioval) {
icculus@1982
   308
                    dst_sample = min_audioval;
icculus@1982
   309
                }
icculus@2014
   310
                *(dst32++) = SDL_SwapFloatBE((float) dst_sample);
icculus@1982
   311
            }
icculus@1982
   312
        }
icculus@1982
   313
        break;
icculus@1982
   314
slouken@1895
   315
    default:                   /* If this happens... FIXME! */
slouken@1895
   316
        SDL_SetError("SDL_MixAudio(): unknown audio format");
slouken@1895
   317
        return;
slouken@1895
   318
    }
slouken@0
   319
}
slouken@574
   320
slouken@1895
   321
/* vi: set ts=4 sw=4 expandtab: */