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view src/audio/SDL_mixer.c @ 3191:91b335df6fc8

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Fixed bug #750 Since many different event structures include windowID it should be placed near the beginning of the structure (preferably right after type) so it's position is the same between different events. This is to avoid code like this: if (event.type == SDL_WINDOWEVENT) win = event.window.windowID; else if ((SDL_EVENTMASK(event.type) & SDL_KEYEVENTMASK) != 0) win = event.key.windowID; else if (event.type == SDL_TEXTINPUT) win = event.text.windowID; else if (event.type == SDL_MOUSEMOTION) win = event.motion.windowID; else if ((SDL_EVENTMASK(event.type) & (SDL_MOUBUTTONDOWNMASK | SDL_MOUBUTTONUPMASK)) != 0) win = event.button.windowID; else if (event.type == SDL_MOUSEWHEEL) win = event.wheel.windowID; ... in favor of: win = event.window.windowID;
author Sam Lantinga <slouken@libsdl.org>
date Wed, 10 Jun 2009 14:00:21 +0000
parents 99210400e8b9
children 4d46850be3f6
line wrap: on
line source

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 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
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library; if not, write to the Free Software
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

    Sam Lantinga
    slouken@libsdl.org
*/
#include "SDL_config.h"

/* This provides the default mixing callback for the SDL audio routines */

#include "SDL_cpuinfo.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "SDL_sysaudio.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_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
                   Uint32 len, int volume)
{
    if (volume == 0) {
        return;
    }

    switch (format) {

    case AUDIO_U8:
        {
#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
            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(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
            if (SDL_HasMMX()) {
                SDL_MixAudio_MMX_S8((char *) dst, (char *) src,
                                    (unsigned int) len, (int) volume);
            } else
#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
            if (SDL_HasMMX()) {
                SDL_MixAudio_MMX_S8_VC((char *) dst, (char *) src,
                                       (unsigned int) len, (int) volume);
            } else
#endif
#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
                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(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
            if (SDL_HasMMX()) {
                SDL_MixAudio_MMX_S16((char *) dst, (char *) src,
                                     (unsigned int) len, (int) volume);
            } else
#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
            if (SDL_HasMMX()) {
                SDL_MixAudio_MMX_S16_VC((char *) dst, (char *) src,
                                        (unsigned int) len, (int) volume);
            } else
#endif
#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
                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(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
            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;

    case AUDIO_S32LSB:
        {
            const Uint32 *src32 = (Uint32 *) src;
            Uint32 *dst32 = (Uint32 *) dst;
            Sint64 src1, src2;
            Sint64 dst_sample;
            const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
            const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));

            len /= 4;
            while (len--) {
                src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
                src32++;
                ADJUST_VOLUME(src1, volume);
                src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
                dst_sample = src1 + src2;
                if (dst_sample > max_audioval) {
                    dst_sample = max_audioval;
                } else if (dst_sample < min_audioval) {
                    dst_sample = min_audioval;
                }
                *(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
            }
        }
        break;

    case AUDIO_S32MSB:
        {
            const Uint32 *src32 = (Uint32 *) src;
            Uint32 *dst32 = (Uint32 *) dst;
            Sint64 src1, src2;
            Sint64 dst_sample;
            const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
            const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));

            len /= 4;
            while (len--) {
                src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
                src32++;
                ADJUST_VOLUME(src1, volume);
                src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
                dst_sample = src1 + src2;
                if (dst_sample > max_audioval) {
                    dst_sample = max_audioval;
                } else if (dst_sample < min_audioval) {
                    dst_sample = min_audioval;
                }
                *(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
            }
        }
        break;

    case AUDIO_F32LSB:
        {
            const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
            const float fvolume = (float) volume;
            const float *src32 = (float *) src;
            float *dst32 = (float *) dst;
            float src1, src2;
            double dst_sample;
            /* !!! FIXME: are these right? */
            const double max_audioval = 3.402823466e+38F;
            const double min_audioval = -3.402823466e+38F;

            len /= 4;
            while (len--) {
                src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
                src2 = SDL_SwapFloatLE(*dst32);
                src32++;

                dst_sample = ((double) src1) + ((double) src2);
                if (dst_sample > max_audioval) {
                    dst_sample = max_audioval;
                } else if (dst_sample < min_audioval) {
                    dst_sample = min_audioval;
                }
                *(dst32++) = SDL_SwapFloatLE((float) dst_sample);
            }
        }
        break;

    case AUDIO_F32MSB:
        {
            const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
            const float fvolume = (float) volume;
            const float *src32 = (float *) src;
            float *dst32 = (float *) dst;
            float src1, src2;
            double dst_sample;
            /* !!! FIXME: are these right? */
            const double max_audioval = 3.402823466e+38F;
            const double min_audioval = -3.402823466e+38F;

            len /= 4;
            while (len--) {
                src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
                src2 = SDL_SwapFloatBE(*dst32);
                src32++;

                dst_sample = ((double) src1) + ((double) src2);
                if (dst_sample > max_audioval) {
                    dst_sample = max_audioval;
                } else if (dst_sample < min_audioval) {
                    dst_sample = min_audioval;
                }
                *(dst32++) = SDL_SwapFloatBE((float) dst_sample);
            }
        }
        break;

    default:                   /* If this happens... FIXME! */
        SDL_SetError("SDL_MixAudio(): unknown audio format");
        return;
    }
}

/* vi: set ts=4 sw=4 expandtab: */