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sdl2_frt/src/audio/SDL_audiocvt.c

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Fixed crash if initialization of EGL failed but was tried again later. The internal function SDL_EGL_LoadLibrary() did not delete and remove a mostly uninitialized data structure if loading the library first failed. A later try to use EGL then skipped initialization and assumed it was previously successful because the data structure now already existed. This led to at least one crash in the internal function SDL_EGL_ChooseConfig() because a NULL pointer was dereferenced to make a call to eglBindAPI().
2015-06-21 17:33:46 +02:00
/*
Simple DirectMedia Layer
Updated copyright to 2016
2016-01-02 19:10:34 +01:00
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Fixed crash if initialization of EGL failed but was tried again later. The internal function SDL_EGL_LoadLibrary() did not delete and remove a mostly uninitialized data structure if loading the library first failed. A later try to use EGL then skipped initialization and assumed it was previously successful because the data structure now already existed. This led to at least one crash in the internal function SDL_EGL_ChooseConfig() because a NULL pointer was dereferenced to make a call to eglBindAPI().
2015-06-21 17:33:46 +02:00
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"
/* Functions for audio drivers to perform runtime conversion of audio format */
#include "SDL_audio.h"
#include "SDL_audio_c.h"
#include "SDL_assert.h"
/* #define DEBUG_CONVERT */
/* Effectively mix right and left channels into a single channel */
static void SDLCALL
SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
Sint32 sample;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to mono\n");
#endif
switch (format & (SDL_AUDIO_MASK_SIGNED |
SDL_AUDIO_MASK_BITSIZE |
SDL_AUDIO_MASK_DATATYPE)) {
case AUDIO_U8:
{
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
for (i = cvt->len_cvt / 2; i; --i) {
sample = src[0] + src[1];
*dst = (Uint8) (sample / 2);
src += 2;
dst += 1;
}
}
break;
case AUDIO_S8:
{
Sint8 *src, *dst;
src = (Sint8 *) cvt->buf;
dst = (Sint8 *) cvt->buf;
for (i = cvt->len_cvt / 2; i; --i) {
sample = src[0] + src[1];
*dst = (Sint8) (sample / 2);
src += 2;
dst += 1;
}
}
break;
case AUDIO_U16:
{
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 4; i; --i) {
sample = (Uint16) ((src[0] << 8) | src[1]) +
(Uint16) ((src[2] << 8) | src[3]);
sample /= 2;
dst[1] = (sample & 0xFF);
sample >>= 8;
dst[0] = (sample & 0xFF);
src += 4;
dst += 2;
}
} else {
for (i = cvt->len_cvt / 4; i; --i) {
sample = (Uint16) ((src[1] << 8) | src[0]) +
(Uint16) ((src[3] << 8) | src[2]);
sample /= 2;
dst[0] = (sample & 0xFF);
sample >>= 8;
dst[1] = (sample & 0xFF);
src += 4;
dst += 2;
}
}
}
break;
case AUDIO_S16:
{
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 4; i; --i) {
sample = (Sint16) ((src[0] << 8) | src[1]) +
(Sint16) ((src[2] << 8) | src[3]);
sample /= 2;
dst[1] = (sample & 0xFF);
sample >>= 8;
dst[0] = (sample & 0xFF);
src += 4;
dst += 2;
}
} else {
for (i = cvt->len_cvt / 4; i; --i) {
sample = (Sint16) ((src[1] << 8) | src[0]) +
(Sint16) ((src[3] << 8) | src[2]);
sample /= 2;
dst[0] = (sample & 0xFF);
sample >>= 8;
dst[1] = (sample & 0xFF);
src += 4;
dst += 2;
}
}
}
break;
case AUDIO_S32:
{
const Uint32 *src = (const Uint32 *) cvt->buf;
Uint32 *dst = (Uint32 *) cvt->buf;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
const Sint64 added =
(((Sint64) (Sint32) SDL_SwapBE32(src[0])) +
((Sint64) (Sint32) SDL_SwapBE32(src[1])));
*(dst++) = SDL_SwapBE32((Uint32) ((Sint32) (added / 2)));
}
} else {
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
const Sint64 added =
(((Sint64) (Sint32) SDL_SwapLE32(src[0])) +
((Sint64) (Sint32) SDL_SwapLE32(src[1])));
*(dst++) = SDL_SwapLE32((Uint32) ((Sint32) (added / 2)));
}
}
}
break;
case AUDIO_F32:
{
const float *src = (const float *) cvt->buf;
float *dst = (float *) cvt->buf;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
const float src1 = SDL_SwapFloatBE(src[0]);
const float src2 = SDL_SwapFloatBE(src[1]);
const double added = ((double) src1) + ((double) src2);
const float halved = (float) (added * 0.5);
*(dst++) = SDL_SwapFloatBE(halved);
}
} else {
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
const float src1 = SDL_SwapFloatLE(src[0]);
const float src2 = SDL_SwapFloatLE(src[1]);
const double added = ((double) src1) + ((double) src2);
const float halved = (float) (added * 0.5);
*(dst++) = SDL_SwapFloatLE(halved);
}
}
}
break;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Discard top 4 channels */
static void SDLCALL
SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting down from 6 channels to stereo\n");
#endif
#define strip_chans_6_to_2(type) \
{ \
const type *src = (const type *) cvt->buf; \
type *dst = (type *) cvt->buf; \
for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
dst[0] = src[0]; \
dst[1] = src[1]; \
src += 6; \
dst += 2; \
} \
}
/* this function only cares about typesize, and data as a block of bits. */
switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
strip_chans_6_to_2(Uint8);
break;
case 16:
strip_chans_6_to_2(Uint16);
break;
case 32:
strip_chans_6_to_2(Uint32);
break;
}
#undef strip_chans_6_to_2
cvt->len_cvt /= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Discard top 2 channels of 6 */
static void SDLCALL
SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting 6 down to quad\n");
#endif
#define strip_chans_6_to_4(type) \
{ \
const type *src = (const type *) cvt->buf; \
type *dst = (type *) cvt->buf; \
for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
dst[0] = src[0]; \
dst[1] = src[1]; \
dst[2] = src[2]; \
dst[3] = src[3]; \
src += 6; \
dst += 4; \
} \
}
/* this function only cares about typesize, and data as a block of bits. */
switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
strip_chans_6_to_4(Uint8);
break;
case 16:
strip_chans_6_to_4(Uint16);
break;
case 32:
strip_chans_6_to_4(Uint32);
break;
}
#undef strip_chans_6_to_4
cvt->len_cvt /= 6;
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Duplicate a mono channel to both stereo channels */
static void SDLCALL
SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to stereo\n");
#endif
#define dup_chans_1_to_2(type) \
{ \
const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
type *dst = (type *) (cvt->buf + cvt->len_cvt * 2); \
for (i = cvt->len_cvt / sizeof(type); i; --i) { \
src -= 1; \
dst -= 2; \
dst[0] = dst[1] = *src; \
} \
}
/* this function only cares about typesize, and data as a block of bits. */
switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
dup_chans_1_to_2(Uint8);
break;
case 16:
dup_chans_1_to_2(Uint16);
break;
case 32:
dup_chans_1_to_2(Uint32);
break;
}
#undef dup_chans_1_to_2
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Duplicate a stereo channel to a pseudo-5.1 stream */
static void SDLCALL
SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting stereo to surround\n");
#endif
switch (format & (SDL_AUDIO_MASK_SIGNED |
SDL_AUDIO_MASK_BITSIZE |
SDL_AUDIO_MASK_DATATYPE)) {
case AUDIO_U8:
{
Uint8 *src, *dst, lf, rf, ce;
src = (Uint8 *) (cvt->buf + cvt->len_cvt);
dst = (Uint8 *) (cvt->buf + cvt->len_cvt * 3);
for (i = cvt->len_cvt; i; --i) {
dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf / 2) + (rf / 2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
dst[4] = ce;
dst[5] = ce;
}
}
break;
case AUDIO_S8:
{
Sint8 *src, *dst, lf, rf, ce;
src = (Sint8 *) cvt->buf + cvt->len_cvt;
dst = (Sint8 *) cvt->buf + cvt->len_cvt * 3;
for (i = cvt->len_cvt; i; --i) {
dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf / 2) + (rf / 2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
dst[4] = ce;
dst[5] = ce;
}
}
break;
case AUDIO_U16:
{
Uint8 *src, *dst;
Uint16 lf, rf, ce, lr, rr;
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 3;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 12;
src -= 4;
lf = (Uint16) ((src[0] << 8) | src[1]);
rf = (Uint16) ((src[2] << 8) | src[3]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf & 0xFF);
dst[0] = ((lf >> 8) & 0xFF);
dst[3] = (rf & 0xFF);
dst[2] = ((rf >> 8) & 0xFF);
dst[1 + 4] = (lr & 0xFF);
dst[0 + 4] = ((lr >> 8) & 0xFF);
dst[3 + 4] = (rr & 0xFF);
dst[2 + 4] = ((rr >> 8) & 0xFF);
dst[1 + 8] = (ce & 0xFF);
dst[0 + 8] = ((ce >> 8) & 0xFF);
dst[3 + 8] = (ce & 0xFF);
dst[2 + 8] = ((ce >> 8) & 0xFF);
}
} else {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 12;
src -= 4;
lf = (Uint16) ((src[1] << 8) | src[0]);
rf = (Uint16) ((src[3] << 8) | src[2]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf & 0xFF);
dst[1] = ((lf >> 8) & 0xFF);
dst[2] = (rf & 0xFF);
dst[3] = ((rf >> 8) & 0xFF);
dst[0 + 4] = (lr & 0xFF);
dst[1 + 4] = ((lr >> 8) & 0xFF);
dst[2 + 4] = (rr & 0xFF);
dst[3 + 4] = ((rr >> 8) & 0xFF);
dst[0 + 8] = (ce & 0xFF);
dst[1 + 8] = ((ce >> 8) & 0xFF);
dst[2 + 8] = (ce & 0xFF);
dst[3 + 8] = ((ce >> 8) & 0xFF);
}
}
}
break;
case AUDIO_S16:
{
Uint8 *src, *dst;
Sint16 lf, rf, ce, lr, rr;
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 3;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 12;
src -= 4;
lf = (Sint16) ((src[0] << 8) | src[1]);
rf = (Sint16) ((src[2] << 8) | src[3]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf & 0xFF);
dst[0] = ((lf >> 8) & 0xFF);
dst[3] = (rf & 0xFF);
dst[2] = ((rf >> 8) & 0xFF);
dst[1 + 4] = (lr & 0xFF);
dst[0 + 4] = ((lr >> 8) & 0xFF);
dst[3 + 4] = (rr & 0xFF);
dst[2 + 4] = ((rr >> 8) & 0xFF);
dst[1 + 8] = (ce & 0xFF);
dst[0 + 8] = ((ce >> 8) & 0xFF);
dst[3 + 8] = (ce & 0xFF);
dst[2 + 8] = ((ce >> 8) & 0xFF);
}
} else {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 12;
src -= 4;
lf = (Sint16) ((src[1] << 8) | src[0]);
rf = (Sint16) ((src[3] << 8) | src[2]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf & 0xFF);
dst[1] = ((lf >> 8) & 0xFF);
dst[2] = (rf & 0xFF);
dst[3] = ((rf >> 8) & 0xFF);
dst[0 + 4] = (lr & 0xFF);
dst[1 + 4] = ((lr >> 8) & 0xFF);
dst[2 + 4] = (rr & 0xFF);
dst[3 + 4] = ((rr >> 8) & 0xFF);
dst[0 + 8] = (ce & 0xFF);
dst[1 + 8] = ((ce >> 8) & 0xFF);
dst[2 + 8] = (ce & 0xFF);
dst[3 + 8] = ((ce >> 8) & 0xFF);
}
}
}
break;
case AUDIO_S32:
{
Sint32 lf, rf, ce;
const Uint32 *src = (const Uint32 *) (cvt->buf + cvt->len_cvt);
Uint32 *dst = (Uint32 *) (cvt->buf + cvt->len_cvt * 3);
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
lf = (Sint32) SDL_SwapBE32(src[0]);
rf = (Sint32) SDL_SwapBE32(src[1]);
ce = (lf / 2) + (rf / 2);
dst[0] = SDL_SwapBE32((Uint32) lf);
dst[1] = SDL_SwapBE32((Uint32) rf);
dst[2] = SDL_SwapBE32((Uint32) (lf - ce));
dst[3] = SDL_SwapBE32((Uint32) (rf - ce));
dst[4] = SDL_SwapBE32((Uint32) ce);
dst[5] = SDL_SwapBE32((Uint32) ce);
}
} else {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
lf = (Sint32) SDL_SwapLE32(src[0]);
rf = (Sint32) SDL_SwapLE32(src[1]);
ce = (lf / 2) + (rf / 2);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapLE32((Uint32) (lf - ce));
dst[3] = SDL_SwapLE32((Uint32) (rf - ce));
dst[4] = SDL_SwapLE32((Uint32) ce);
dst[5] = SDL_SwapLE32((Uint32) ce);
}
}
}
break;
case AUDIO_F32:
{
float lf, rf, ce;
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
lf = SDL_SwapFloatBE(src[0]);
rf = SDL_SwapFloatBE(src[1]);
ce = (lf * 0.5f) + (rf * 0.5f);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapFloatBE(lf - ce);
dst[3] = SDL_SwapFloatBE(rf - ce);
dst[4] = dst[5] = SDL_SwapFloatBE(ce);
}
} else {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
lf = SDL_SwapFloatLE(src[0]);
rf = SDL_SwapFloatLE(src[1]);
ce = (lf * 0.5f) + (rf * 0.5f);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapFloatLE(lf - ce);
dst[3] = SDL_SwapFloatLE(rf - ce);
dst[4] = dst[5] = SDL_SwapFloatLE(ce);
}
}
}
break;
}
cvt->len_cvt *= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Duplicate a stereo channel to a pseudo-4.0 stream */
static void SDLCALL
SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting stereo to quad\n");
#endif
switch (format & (SDL_AUDIO_MASK_SIGNED |
SDL_AUDIO_MASK_BITSIZE |
SDL_AUDIO_MASK_DATATYPE)) {
case AUDIO_U8:
{
Uint8 *src, *dst, lf, rf, ce;
src = (Uint8 *) (cvt->buf + cvt->len_cvt);
dst = (Uint8 *) (cvt->buf + cvt->len_cvt * 2);
for (i = cvt->len_cvt; i; --i) {
dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf / 2) + (rf / 2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
}
}
break;
case AUDIO_S8:
{
Sint8 *src, *dst, lf, rf, ce;
src = (Sint8 *) cvt->buf + cvt->len_cvt;
dst = (Sint8 *) cvt->buf + cvt->len_cvt * 2;
for (i = cvt->len_cvt; i; --i) {
dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf / 2) + (rf / 2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
}
}
break;
case AUDIO_U16:
{
Uint8 *src, *dst;
Uint16 lf, rf, ce, lr, rr;
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 2;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 8;
src -= 4;
lf = (Uint16) ((src[0] << 8) | src[1]);
rf = (Uint16) ((src[2] << 8) | src[3]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf & 0xFF);
dst[0] = ((lf >> 8) & 0xFF);
dst[3] = (rf & 0xFF);
dst[2] = ((rf >> 8) & 0xFF);
dst[1 + 4] = (lr & 0xFF);
dst[0 + 4] = ((lr >> 8) & 0xFF);
dst[3 + 4] = (rr & 0xFF);
dst[2 + 4] = ((rr >> 8) & 0xFF);
}
} else {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 8;
src -= 4;
lf = (Uint16) ((src[1] << 8) | src[0]);
rf = (Uint16) ((src[3] << 8) | src[2]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf & 0xFF);
dst[1] = ((lf >> 8) & 0xFF);
dst[2] = (rf & 0xFF);
dst[3] = ((rf >> 8) & 0xFF);
dst[0 + 4] = (lr & 0xFF);
dst[1 + 4] = ((lr >> 8) & 0xFF);
dst[2 + 4] = (rr & 0xFF);
dst[3 + 4] = ((rr >> 8) & 0xFF);
}
}
}
break;
case AUDIO_S16:
{
Uint8 *src, *dst;
Sint16 lf, rf, ce, lr, rr;
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 2;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 8;
src -= 4;
lf = (Sint16) ((src[0] << 8) | src[1]);
rf = (Sint16) ((src[2] << 8) | src[3]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf & 0xFF);
dst[0] = ((lf >> 8) & 0xFF);
dst[3] = (rf & 0xFF);
dst[2] = ((rf >> 8) & 0xFF);
dst[1 + 4] = (lr & 0xFF);
dst[0 + 4] = ((lr >> 8) & 0xFF);
dst[3 + 4] = (rr & 0xFF);
dst[2 + 4] = ((rr >> 8) & 0xFF);
}
} else {
for (i = cvt->len_cvt / 4; i; --i) {
dst -= 8;
src -= 4;
lf = (Sint16) ((src[1] << 8) | src[0]);
rf = (Sint16) ((src[3] << 8) | src[2]);
ce = (lf / 2) + (rf / 2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf & 0xFF);
dst[1] = ((lf >> 8) & 0xFF);
dst[2] = (rf & 0xFF);
dst[3] = ((rf >> 8) & 0xFF);
dst[0 + 4] = (lr & 0xFF);
dst[1 + 4] = ((lr >> 8) & 0xFF);
dst[2 + 4] = (rr & 0xFF);
dst[3 + 4] = ((rr >> 8) & 0xFF);
}
}
}
break;
case AUDIO_S32:
{
const Uint32 *src = (const Uint32 *) (cvt->buf + cvt->len_cvt);
Uint32 *dst = (Uint32 *) (cvt->buf + cvt->len_cvt * 2);
Sint32 lf, rf, ce;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 4;
src -= 2;
lf = (Sint32) SDL_SwapBE32(src[0]);
rf = (Sint32) SDL_SwapBE32(src[1]);
ce = (lf / 2) + (rf / 2);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapBE32((Uint32) (lf - ce));
dst[3] = SDL_SwapBE32((Uint32) (rf - ce));
}
} else {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 4;
src -= 2;
lf = (Sint32) SDL_SwapLE32(src[0]);
rf = (Sint32) SDL_SwapLE32(src[1]);
ce = (lf / 2) + (rf / 2);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapLE32((Uint32) (lf - ce));
dst[3] = SDL_SwapLE32((Uint32) (rf - ce));
}
}
}
break;
case AUDIO_F32:
{
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
float lf, rf, ce;
if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 4;
src -= 2;
lf = SDL_SwapFloatBE(src[0]);
rf = SDL_SwapFloatBE(src[1]);
ce = (lf / 2) + (rf / 2);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapFloatBE(lf - ce);
dst[3] = SDL_SwapFloatBE(rf - ce);
}
} else {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 4;
src -= 2;
lf = SDL_SwapFloatLE(src[0]);
rf = SDL_SwapFloatLE(src[1]);
ce = (lf / 2) + (rf / 2);
dst[0] = src[0];
dst[1] = src[1];
dst[2] = SDL_SwapFloatLE(lf - ce);
dst[3] = SDL_SwapFloatLE(rf - ce);
}
}
}
break;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
int
SDL_ConvertAudio(SDL_AudioCVT * cvt)
{
/* !!! FIXME: (cvt) should be const; stack-copy it here. */
/* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
/* Make sure there's data to convert */
if (cvt->buf == NULL) {
SDL_SetError("No buffer allocated for conversion");
return (-1);
}
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if (cvt->filters[0] == NULL) {
return (0);
}
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0] (cvt, cvt->src_format);
return (0);
}
static SDL_AudioFilter
SDL_HandTunedTypeCVT(SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt)
{
/*
* Fill in any future conversions that are specialized to a
* processor, platform, compiler, or library here.
*/
return NULL; /* no specialized converter code available. */
}
/*
* Find a converter between two data types. We try to select a hand-tuned
* asm/vectorized/optimized function first, and then fallback to an
* autogenerated function that is customized to convert between two
* specific data types.
*/
static int
SDL_BuildAudioTypeCVT(SDL_AudioCVT * cvt,
SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt)
{
if (src_fmt != dst_fmt) {
const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
SDL_AudioFilter filter = SDL_HandTunedTypeCVT(src_fmt, dst_fmt);
/* No hand-tuned converter? Try the autogenerated ones. */
if (filter == NULL) {
int i;
for (i = 0; sdl_audio_type_filters[i].filter != NULL; i++) {
const SDL_AudioTypeFilters *filt = &sdl_audio_type_filters[i];
if ((filt->src_fmt == src_fmt) && (filt->dst_fmt == dst_fmt)) {
filter = filt->filter;
break;
}
}
if (filter == NULL) {
SDL_SetError("No conversion available for these formats");
return -1;
}
}
/* Update (cvt) with filter details... */
cvt->filters[cvt->filter_index++] = filter;
if (src_bitsize < dst_bitsize) {
const int mult = (dst_bitsize / src_bitsize);
cvt->len_mult *= mult;
cvt->len_ratio *= mult;
} else if (src_bitsize > dst_bitsize) {
cvt->len_ratio /= (src_bitsize / dst_bitsize);
}
return 1; /* added a converter. */
}
return 0; /* no conversion necessary. */
}
static SDL_AudioFilter
SDL_HandTunedResampleCVT(SDL_AudioCVT * cvt, int dst_channels,
int src_rate, int dst_rate)
{
/*
* Fill in any future conversions that are specialized to a
* processor, platform, compiler, or library here.
*/
return NULL; /* no specialized converter code available. */
}
static int
SDL_FindFrequencyMultiple(const int src_rate, const int dst_rate)
{
int retval = 0;
/* If we only built with the arbitrary resamplers, ignore multiples. */
#if !LESS_RESAMPLERS
int lo, hi;
int div;
SDL_assert(src_rate != 0);
SDL_assert(dst_rate != 0);
SDL_assert(src_rate != dst_rate);
if (src_rate < dst_rate) {
lo = src_rate;
hi = dst_rate;
} else {
lo = dst_rate;
hi = src_rate;
}
/* zero means "not a supported multiple" ... we only do 2x and 4x. */
if ((hi % lo) != 0)
return 0; /* not a multiple. */
div = hi / lo;
retval = ((div == 2) || (div == 4)) ? div : 0;
#endif
return retval;
}
static int
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, int dst_channels,
int src_rate, int dst_rate)
{
if (src_rate != dst_rate) {
SDL_AudioFilter filter = SDL_HandTunedResampleCVT(cvt, dst_channels,
src_rate, dst_rate);
/* No hand-tuned converter? Try the autogenerated ones. */
if (filter == NULL) {
int i;
const int upsample = (src_rate < dst_rate) ? 1 : 0;
const int multiple =
SDL_FindFrequencyMultiple(src_rate, dst_rate);
for (i = 0; sdl_audio_rate_filters[i].filter != NULL; i++) {
const SDL_AudioRateFilters *filt = &sdl_audio_rate_filters[i];
if ((filt->fmt == cvt->dst_format) &&
(filt->channels == dst_channels) &&
(filt->upsample == upsample) &&
(filt->multiple == multiple)) {
filter = filt->filter;
break;
}
}
if (filter == NULL) {
SDL_SetError("No conversion available for these rates");
return -1;
}
}
/* Update (cvt) with filter details... */
cvt->filters[cvt->filter_index++] = filter;
if (src_rate < dst_rate) {
const double mult = ((double) dst_rate) / ((double) src_rate);
cvt->len_mult *= (int) SDL_ceil(mult);
cvt->len_ratio *= mult;
} else {
cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
}
return 1; /* added a converter. */
}
return 0; /* no conversion necessary. */
}
/* Creates a set of audio filters to convert from one format to another.
Returns -1 if the format conversion is not supported, 0 if there's
no conversion needed, or 1 if the audio filter is set up.
*/
int
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
{
/*
* !!! FIXME: reorder filters based on which grow/shrink the buffer.
* !!! FIXME: ideally, we should do everything that shrinks the buffer
* !!! FIXME: first, so we don't have to process as many bytes in a given
* !!! FIXME: filter and abuse the CPU cache less. This might not be as
* !!! FIXME: good in practice as it sounds in theory, though.
*/
/* Sanity check target pointer */
if (cvt == NULL) {
return SDL_InvalidParamError("cvt");
}
/* there are no unsigned types over 16 bits, so catch this up front. */
if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
return SDL_SetError("Invalid source format");
}
if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
return SDL_SetError("Invalid destination format");
}
/* prevent possible divisions by zero, etc. */
if ((src_channels == 0) || (dst_channels == 0)) {
return SDL_SetError("Source or destination channels is zero");
}
if ((src_rate == 0) || (dst_rate == 0)) {
return SDL_SetError("Source or destination rate is zero");
}
#ifdef DEBUG_CONVERT
printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
#endif
/* Start off with no conversion necessary */
SDL_zerop(cvt);
cvt->src_format = src_fmt;
cvt->dst_format = dst_fmt;
cvt->needed = 0;
cvt->filter_index = 0;
cvt->filters[0] = NULL;
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
/* Convert data types, if necessary. Updates (cvt). */
if (SDL_BuildAudioTypeCVT(cvt, src_fmt, dst_fmt) == -1) {
return -1; /* shouldn't happen, but just in case... */
}
/* Channel conversion */
if (src_channels != dst_channels) {
if ((src_channels == 1) && (dst_channels > 1)) {
cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
cvt->len_mult *= 2;
src_channels = 2;
cvt->len_ratio *= 2;
}
if ((src_channels == 2) && (dst_channels == 6)) {
cvt->filters[cvt->filter_index++] = SDL_ConvertSurround;
src_channels = 6;
cvt->len_mult *= 3;
cvt->len_ratio *= 3;
}
if ((src_channels == 2) && (dst_channels == 4)) {
cvt->filters[cvt->filter_index++] = SDL_ConvertSurround_4;
src_channels = 4;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
}
while ((src_channels * 2) <= dst_channels) {
cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
cvt->len_mult *= 2;
src_channels *= 2;
cvt->len_ratio *= 2;
}
if ((src_channels == 6) && (dst_channels <= 2)) {
cvt->filters[cvt->filter_index++] = SDL_ConvertStrip;
src_channels = 2;
cvt->len_ratio /= 3;
}
if ((src_channels == 6) && (dst_channels == 4)) {
cvt->filters[cvt->filter_index++] = SDL_ConvertStrip_2;
src_channels = 4;
cvt->len_ratio /= 2;
}
/* This assumes that 4 channel audio is in the format:
Left {front/back} + Right {front/back}
so converting to L/R stereo works properly.
*/
while (((src_channels % 2) == 0) &&
((src_channels / 2) >= dst_channels)) {
cvt->filters[cvt->filter_index++] = SDL_ConvertMono;
src_channels /= 2;
cvt->len_ratio /= 2;
}
if (src_channels != dst_channels) {
/* Uh oh.. */ ;
}
}
/* Do rate conversion, if necessary. Updates (cvt). */
if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) ==
-1) {
return -1; /* shouldn't happen, but just in case... */
}
/* Set up the filter information */
if (cvt->filter_index != 0) {
cvt->needed = 1;
cvt->src_format = src_fmt;
cvt->dst_format = dst_fmt;
cvt->len = 0;
cvt->buf = NULL;
cvt->filters[cvt->filter_index] = NULL;
}
return (cvt->needed);
}
/* vi: set ts=4 sw=4 expandtab: */
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