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cave-story-solaris/external/SDL2/src/audio/SDL_mixer.c
Clownacy ac465d29b4 Mean CMake dependency overhaul 
I'm taking a page from Dolphin's book, and including copies of each
dependency's source code. This combines the ease of use of including
pre-built libraries instead of needing to navigate a package manager
 - as is (or was) the case for MSVC - with the portability of using
packages. Granted, this method's more of a jack of all trades,
master of none, since it's *less* user-friendly than prebuilt
packages (compilation times), and you don't get the per-distro
compatibility fixes you'd get from a package manager.

You can still use system libs if you want. In fact, it's still the
default behaviour: compiling the libs manually is just a fallback.
I'll add an option to force-enable this soon, however, since it's a
nicer way to produce static MSYS2 builds than the hackish nightmare
that I was using before. Not to mention, having my own copy of the
sources means I can provide my own fixes and tweaks your package
manager may not. For example, I can combine MSYS2's FreeType
subpixel rendering with vcpkg's fix for SDL2 exporting its symbols
in static builds.
2019-04-26 01:52:02 +01:00

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/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
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_cpuinfo.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#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)
* 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(__mcoldfire__) && 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:
{
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;
}
}
break;
case AUDIO_S16LSB:
{
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;
}
}
break;
case AUDIO_S16MSB:
{
#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && 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_U16LSB:
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = 0xFFFF;
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;
}
dst[0] = dst_sample & 0xFF;
dst_sample >>= 8;
dst[1] = dst_sample & 0xFF;
dst += 2;
}
}
break;
case AUDIO_U16MSB:
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = 0xFFFF;
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;
}
dst[1] = dst_sample & 0xFF;
dst_sample >>= 8;
dst[0] = dst_sample & 0xFF;
dst += 2;
}
}
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_MixAudioFormat(): unknown audio format");
return;
}
}
/* vi: set ts=4 sw=4 expandtab: */
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