/*

Simple DirectMedia Layer

Copyright (C) 1997-2017 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"

#include "SDL_audio.h"

#include "SDL_audio_c.h"

#include "SDL_assert.h"

#define DIVBY127 0.0078740157480315f

#define DIVBY32767 3.05185094759972e-05f

#define DIVBY2147483647 4.6566128752458e-10f

void SDLCALL

SDL_Convert_S8_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const Uint8 *src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;

float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;

int i;

LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32");

for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) {

*dst = (((float) ((Sint8) *src)) * DIVBY127);

}

cvt->len_cvt *= 4;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void SDLCALL

SDL_Convert_U8_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const Uint8 *src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;

float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;

int i;

LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32");

for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) {

*dst = ((((float) *src) * DIVBY127) - 1.0f);

}

cvt->len_cvt *= 4;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void SDLCALL

SDL_Convert_S16_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const Sint16 *src = ((const Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;

float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;

int i;

LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32");

for (i = cvt->len_cvt / sizeof (Sint16); i; --i, --src, --dst) {

*dst = (((float) *src) * DIVBY32767);

}

cvt->len_cvt *= 2;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void SDLCALL

SDL_Convert_U16_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const Uint16 *src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;

float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;

int i;

LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32");

for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) {

*dst = ((((float) *src) * DIVBY32767) - 1.0f);

}

cvt->len_cvt *= 2;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void SDLCALL

SDL_Convert_S32_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const Uint32 *src = (const Uint32 *) cvt->buf;

float *dst = (float *) cvt->buf;

int i;

LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32");

for (i = cvt->len_cvt / sizeof (Sint32); i; --i, ++src, ++dst) {

*dst = (((float) *src) * DIVBY2147483647);

}

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void SDLCALL

SDL_Convert_F32_to_S8(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const float *src = (const float *) cvt->buf;

Sint8 *dst = (Sint8 *) cvt->buf;

int i;

LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8");

for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {

*dst = (Sint8) (*src * 127.0f);

}

cvt->len_cvt /= 4;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_S8);

}

}

void SDLCALL

SDL_Convert_F32_to_U8(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const float *src = (const float *) cvt->buf;

Uint8 *dst = (Uint8 *) cvt->buf;

int i;

LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8");

for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {

*dst = (Uint8) ((*src + 1.0f) * 127.0f);

}

cvt->len_cvt /= 4;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_U8);

}

}

void SDLCALL

SDL_Convert_F32_to_S16(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const float *src = (const float *) cvt->buf;

Sint16 *dst = (Sint16 *) cvt->buf;

int i;

LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16");

for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {

*dst = (Sint16) (*src * 32767.0f);

}

cvt->len_cvt /= 2;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS);

}

}

void SDLCALL

SDL_Convert_F32_to_U16(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const float *src = (const float *) cvt->buf;

Uint16 *dst = (Uint16 *) cvt->buf;

int i;

LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16");

for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {

*dst = (Uint16) ((*src + 1.0f) * 32767.0f);

}

cvt->len_cvt /= 2;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS);

}

}

void SDLCALL

SDL_Convert_F32_to_S32(SDL_AudioCVT *cvt, SDL_AudioFormat format)

{

const float *src = (const float *) cvt->buf;

Sint32 *dst = (Sint32 *) cvt->buf;

int i;

LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32");

for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {

*dst = (Sint32) (*src * 2147483647.0);

}

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS);

}

}

void

SDL_Upsample_Arbitrary(SDL_AudioCVT *cvt, const int channels)

{

const int srcsize = cvt->len_cvt - (64 * channels);

const int dstsize = (int) (((double)(cvt->len_cvt/(channels*4))) * cvt->rate_incr) * (channels*4);

register int eps = 0;

float *dst = ((float *) (cvt->buf + dstsize)) - channels;

const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - channels;

const float *target = ((const float *) cvt->buf);

const size_t cpy = sizeof (float) * channels;

float last_sample[8];

float sample[8];

int i;

#if DEBUG_CONVERT

fprintf(stderr, "Upsample arbitrary (x%f), %d channels.\n", cvt->rate_incr, channels);

#endif

SDL_assert(channels <= 8);

SDL_memcpy(sample, src, cpy);

SDL_memcpy(last_sample, src, cpy);

while (dst > target) {

SDL_memcpy(dst, sample, cpy);

dst -= channels;

eps += srcsize;

if ((eps << 1) >= dstsize) {

src -= channels;

for (i = 0; i < channels; i++) {

sample[i] = (float) ((((double) src[i]) + ((double) last_sample[i])) * 0.5);

}

SDL_memcpy(last_sample, sample, cpy);

eps -= dstsize;

}

}

cvt->len_cvt = dstsize;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void

SDL_Downsample_Arbitrary(SDL_AudioCVT *cvt, const int channels)

{

const int srcsize = cvt->len_cvt - (64 * channels);

const int dstsize = (int) (((double)(cvt->len_cvt/(channels*4))) * cvt->rate_incr) * (channels*4);

register int eps = 0;

float *dst = (float *) cvt->buf;

const float *src = (float *) cvt->buf;

const float *target = (const float *) (cvt->buf + dstsize);

const size_t cpy = sizeof (float) * channels;

float last_sample[8];

float sample[8];

int i;

#if DEBUG_CONVERT

fprintf(stderr, "Downsample arbitrary (x%f), %d channels.\n", cvt->rate_incr, channels);

#endif

SDL_assert(channels <= 8);

SDL_memcpy(sample, src, cpy);

SDL_memcpy(last_sample, src, cpy);

while (dst < target) {

src += channels;

eps += dstsize;

if ((eps << 1) >= srcsize) {

SDL_memcpy(dst, sample, cpy);

dst += channels;

for (i = 0; i < channels; i++) {

sample[i] = (float) ((((double) src[i]) + ((double) last_sample[i])) * 0.5);

}

SDL_memcpy(last_sample, sample, cpy);

eps -= srcsize;

}

}

cvt->len_cvt = dstsize;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void

SDL_Upsample_x2(SDL_AudioCVT *cvt, const int channels)

{

const int dstsize = cvt->len_cvt * 2;

float *dst = ((float *) (cvt->buf + dstsize)) - (channels * 2);

const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - channels;

const float *target = ((const float *) cvt->buf);

const size_t cpy = sizeof (float) * channels;

float last_sample[8];

int i;

#if DEBUG_CONVERT

fprintf(stderr, "Upsample (x2), %d channels.\n", channels);

#endif

SDL_assert(channels <= 8);

SDL_memcpy(last_sample, src, cpy);

while (dst > target) {

for (i = 0; i < channels; i++) {

dst[i] = (float) ((((double)src[i]) + ((double)last_sample[i])) * 0.5);

}

dst -= channels;

SDL_memcpy(dst, src, cpy);

SDL_memcpy(last_sample, src, cpy);

src -= channels;

dst -= channels;

}

cvt->len_cvt = dstsize;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void

SDL_Upsample_x4(SDL_AudioCVT *cvt, const int channels)

{

const int dstsize = cvt->len_cvt * 4;

float *dst = ((float *) (cvt->buf + dstsize)) - (channels * 4);

const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - channels;

const float *target = ((const float *) cvt->buf);

const size_t cpy = sizeof (float) * channels;

float last_sample[8];

int i;

#if DEBUG_CONVERT

fprintf(stderr, "Upsample (x4), %d channels.\n", channels);

#endif

SDL_assert(channels <= 8);

SDL_memcpy(last_sample, src, cpy);

while (dst > target) {

for (i = 0; i < channels; i++) {

dst[i] = (float) ((((double) src[i]) + (3.0 * ((double) last_sample[i]))) * 0.25);

}

dst -= channels;

for (i = 0; i < channels; i++) {

dst[i] = (float) ((((double) src[i]) + ((double) last_sample[i])) * 0.25);

}

dst -= channels;

for (i = 0; i < channels; i++) {

dst[i] = (float) (((3.0 * ((double) src[i])) + ((double) last_sample[i])) * 0.25);

}

dst -= channels;

SDL_memcpy(dst, src, cpy);

dst -= channels;

SDL_memcpy(last_sample, src, cpy);

src -= channels;

}

cvt->len_cvt = dstsize;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

void

SDL_Downsample_Multiple(SDL_AudioCVT *cvt, const int multiple, const int channels)

{

const int dstsize = cvt->len_cvt / multiple;

float *dst = (float *) cvt->buf;

const float *src = (float *) cvt->buf;

const float *target = (const float *) (cvt->buf + dstsize);

const size_t cpy = sizeof (float) * channels;

float last_sample[8];

int i;

#if DEBUG_CONVERT

fprintf(stderr, "Downsample (x%d), %d channels.\n", multiple, channels);

#endif

SDL_assert(channels <= 8);

SDL_memcpy(last_sample, src, cpy);

while (dst < target) {

for (i = 0; i < channels; i++) {

dst[i] = (float) ((((double)src[i]) + ((double)last_sample[i])) * 0.5);

}

dst += channels;

SDL_memcpy(last_sample, src, cpy);

src += (channels * multiple);

}

cvt->len_cvt = dstsize;

if (cvt->filters[++cvt->filter_index]) {

cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);

}

}

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