SDL/src/audio/SDL_mixer.c at SDL_assume · devnexen/SDL

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Simple DirectMedia Layer

This software is provided 'as-is', without any express or implied

warranty. In no event will the authors be held liable for any damages

arising from the use of this software.

Permission is granted to anyone to use this software for any purpose,

including commercial applications, and to alter it and redistribute it

freely, subject to the following restrictions:

1. The origin of this software must not be misrepresented; you must not

claim that you wrote the original software. If you use this software

in a product, an acknowledgment in the product documentation would be

appreciated but is not required.

2. Altered source versions must be plainly marked as such, and must not be

misrepresented as being the original software.

3. This notice may not be removed or altered from any source distribution.

#include "SDL_internal.h"

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

#include "SDL_sysaudio.h"

/* This table is used to add two sound values together and pin

* the value to avoid overflow. (used with permission from ARDI)

static const Uint8 mix8[] = {

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, 0xFF, 0xFF,

0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,

0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF

};

// The volume ranges from 0 - 128

#define ADJUST_VOLUME(type, s, v) ((s) = (type)(((s) * (v)) / SDL_MIX_MAXVOLUME))

#define ADJUST_VOLUME_U8(s, v) ((s) = (Uint8)(((((s) - 128) * (v)) / SDL_MIX_MAXVOLUME) + 128))

// !!! FIXME: this needs some SIMD magic.

int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,

Uint32 len, int volume)

{

if (volume == 0) {

return 0;

}

switch (format) {

case SDL_AUDIO_U8:

{

Uint8 src_sample;

while (len--) {

src_sample = *src;

ADJUST_VOLUME_U8(src_sample, volume);

*dst = mix8[*dst + src_sample];

++dst;

++src;

}

} break;

case SDL_AUDIO_S8:

{

Sint8 *dst8, *src8;

Sint8 src_sample;

int dst_sample;

const int max_audioval = SDL_MAX_SINT8;

const int min_audioval = SDL_MIN_SINT8;

src8 = (Sint8 *)src;

dst8 = (Sint8 *)dst;

while (len--) {

src_sample = *src8;

ADJUST_VOLUME(Sint8, src_sample, volume);

dst_sample = *dst8 + src_sample;

if (dst_sample > max_audioval) {

dst_sample = max_audioval;

} else if (dst_sample < min_audioval) {

dst_sample = min_audioval;

}

*dst8 = (Sint8)dst_sample;

++dst8;

++src8;

}

} break;

case SDL_AUDIO_S16LE:

{

Sint16 src1, src2;

int dst_sample;

const int max_audioval = SDL_MAX_SINT16;

const int min_audioval = SDL_MIN_SINT16;

len /= 2;

while (len--) {

src1 = SDL_SwapLE16(*(Sint16 *)src);

ADJUST_VOLUME(Sint16, src1, volume);

src2 = SDL_SwapLE16(*(Sint16 *)dst);

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;

}

*(Sint16 *)dst = SDL_SwapLE16((Sint16)dst_sample);

dst += 2;

}

} break;

case SDL_AUDIO_S16BE:

{

Sint16 src1, src2;

int dst_sample;

const int max_audioval = SDL_MAX_SINT16;

const int min_audioval = SDL_MIN_SINT16;

len /= 2;

while (len--) {

src1 = SDL_SwapBE16(*(Sint16 *)src);

ADJUST_VOLUME(Sint16, src1, volume);

src2 = SDL_SwapBE16(*(Sint16 *)dst);

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;

}

*(Sint16 *)dst = SDL_SwapBE16((Sint16)dst_sample);

dst += 2;

}

} break;

case SDL_AUDIO_S32LE:

{

const Uint32 *src32 = (Uint32 *)src;

Uint32 *dst32 = (Uint32 *)dst;

Sint64 src1, src2;

Sint64 dst_sample;

const Sint64 max_audioval = SDL_MAX_SINT32;

const Sint64 min_audioval = SDL_MIN_SINT32;

len /= 4;

while (len--) {

src1 = (Sint64)((Sint32)SDL_SwapLE32(*src32));

src32++;

ADJUST_VOLUME(Sint64, 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 SDL_AUDIO_S32BE:

{

const Uint32 *src32 = (Uint32 *)src;

Uint32 *dst32 = (Uint32 *)dst;

Sint64 src1, src2;

Sint64 dst_sample;

const Sint64 max_audioval = SDL_MAX_SINT32;

const Sint64 min_audioval = SDL_MIN_SINT32;

len /= 4;

while (len--) {

src1 = (Sint64)((Sint32)SDL_SwapBE32(*src32));

src32++;

ADJUST_VOLUME(Sint64, 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 SDL_AUDIO_F32LE:

{

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 SDL_AUDIO_F32BE:

{

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!

return SDL_SetError("SDL_MixAudioFormat(): unknown audio format");

}

return 0;

}

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