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SDL_audiocvt.c
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/*
SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2006 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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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
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Lesser General Public License for more details.
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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
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Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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#include <math.h>
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/* Functions for audio drivers to perform runtime conversion of audio format */
#include "SDL_audio.h"
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#include "SDL_audio_c.h"
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#define DEBUG_CONVERT
/* These are fractional multiplication routines. That is, their inputs
are two numbers in the range [-1, 1) and the result falls in that
same range. The output is the same size as the inputs, i.e.
32-bit x 32-bit = 32-bit.
*/
/* We hope here that the right shift includes sign extension */
#ifdef SDL_HAS_64BIT_Type
#define SDL_FixMpy32(a, b) ((((Sint64)a * (Sint64)b) >> 31) & 0xffffffff)
#else
/* If we don't have the 64-bit type, do something more complicated. See http://www.8052.com/mul16.phtml or http://www.cs.uaf.edu/~cs301/notes/Chapter5/node5.html */
#define SDL_FixMpy32(a, b) ((((Sint64)a * (Sint64)b) >> 31) & 0xffffffff)
#endif
#define SDL_FixMpy16(a, b) ((((Sint32)a * (Sint32)b) >> 14) & 0xffff)
#define SDL_FixMpy8(a, b) ((((Sint16)a * (Sint16)b) >> 7) & 0xff)
/* This macro just makes the floating point filtering code not have to be a special case. */
#define SDL_FloatMpy(a, b) (a * b)
/* These macros take floating point numbers in the range [-1.0f, 1.0f) and
represent them as fixed-point numbers in that same range. There's no
checking that the floating point argument is inside the appropriate range.
*/
#define SDL_Make_1_7(a) (Sint8)(a * 128.0f)
#define SDL_Make_1_15(a) (Sint16)(a * 32768.0f)
#define SDL_Make_1_31(a) (Sint32)(a * 2147483648.0f)
#define SDL_Make_2_6(a) (Sint8)(a * 64.0f)
#define SDL_Make_2_14(a) (Sint16)(a * 16384.0f)
#define SDL_Make_2_30(a) (Sint32)(a * 1073741824.0f)
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/* Effectively mix right and left channels into a single channel */
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static void SDLCALL
SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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int i;
Sint32 sample;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to mono\n");
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#endif
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switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
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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];
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*dst = (Uint8) (sample / 2);
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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];
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*dst = (Sint8) (sample / 2);
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src += 2;
dst += 1;
}
}
break;
case AUDIO_U16:
{
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
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for (i = cvt->len_cvt / 4; i; --i) {
sample = (Uint16) ((src[0] << 8) | src[1]) +
(Uint16) ((src[2] << 8) | src[3]);
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sample /= 2;
dst[1] = (sample & 0xFF);
sample >>= 8;
dst[0] = (sample & 0xFF);
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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]);
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sample /= 2;
dst[0] = (sample & 0xFF);
sample >>= 8;
dst[1] = (sample & 0xFF);
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src += 4;
dst += 2;
}
}
}
break;
case AUDIO_S16:
{
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
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for (i = cvt->len_cvt / 4; i; --i) {
sample = (Sint16) ((src[0] << 8) | src[1]) +
(Sint16) ((src[2] << 8) | src[3]);
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sample /= 2;
dst[1] = (sample & 0xFF);
sample >>= 8;
dst[0] = (sample & 0xFF);
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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]);
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sample /= 2;
dst[0] = (sample & 0xFF);
sample >>= 8;
dst[1] = (sample & 0xFF);
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src += 4;
dst += 2;
}
}
}
break;
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case AUDIO_S32:
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{
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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 =
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(((Sint64) (Sint32) SDL_SwapBE32(src[0])) +
((Sint64) (Sint32) SDL_SwapBE32(src[1])));
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*(dst++) = SDL_SwapBE32((Uint32) ((Sint32) (added / 2)));
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}
} else {
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for (i = cvt->len_cvt / 8; i; --i, src += 2) {
const Sint64 added =
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(((Sint64) (Sint32) SDL_SwapLE32(src[0])) +
((Sint64) (Sint32) SDL_SwapLE32(src[1])));
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*(dst++) = SDL_SwapLE32((Uint32) ((Sint32) (added / 2)));
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}
}
}
break;
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case AUDIO_F32:
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{
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const float *src = (const float *) cvt->buf;
float *dst = (float *) cvt->buf;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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const float src1 = SDL_SwapFloatBE(src[0]);
const float src2 = SDL_SwapFloatBE(src[1]);
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const double added = ((double) src1) + ((double) src2);
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const float halved = (float) (added * 0.5);
*(dst++) = SDL_SwapFloatBE(halved);
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}
} else {
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for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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const float src1 = SDL_SwapFloatLE(src[0]);
const float src2 = SDL_SwapFloatLE(src[1]);
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const double added = ((double) src1) + ((double) src2);
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const float halved = (float) (added * 0.5);
*(dst++) = SDL_SwapFloatLE(halved);
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}
}
}
break;
}
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cvt->len_cvt /= 2;
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if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
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}
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/* Discard top 4 channels */
static void SDLCALL
SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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int i;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting down from 6 channels to stereo\n");
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#endif
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#define strip_chans_6_to_2(type) \
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{ \
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; \
} \
}
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/* this function only cares about typesize, and data as a block of bits. */
switch (SDL_AUDIO_BITSIZE(format)) {
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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;
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}
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#undef strip_chans_6_to_2
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cvt->len_cvt /= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
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/* Discard top 2 channels of 6 */
static void SDLCALL
SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
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#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting 6 down to quad\n");
#endif
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#define strip_chans_6_to_4(type) \
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{ \
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; \
} \
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}
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/* this function only cares about typesize, and data as a block of bits. */
switch (SDL_AUDIO_BITSIZE(format)) {
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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;
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}
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#undef strip_chans_6_to_4
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cvt->len_cvt /= 6;
cvt->len_cvt *= 4;
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if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
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}
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/* Duplicate a mono channel to both stereo channels */
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static void SDLCALL
SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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int i;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to stereo\n");
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#endif
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#define dup_chans_1_to_2(type) \
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{ \
const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
type *dst = (type *) (cvt->buf + cvt->len_cvt * 2); \
for (i = cvt->len_cvt / 2; i; --i, --src) { \
const type val = *src; \
dst -= 2; \
dst[0] = dst[1] = val; \
} \
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}
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/* this function only cares about typesize, and data as a block of bits. */
switch (SDL_AUDIO_BITSIZE(format)) {
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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;
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}
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#undef dup_chans_1_to_2
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cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
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}
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/* Duplicate a stereo channel to a pseudo-5.1 stream */
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static void SDLCALL
SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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int i;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting stereo to surround\n");
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#endif
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switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
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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;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
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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;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
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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;
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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;
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const float *src = (const float *) cvt->buf + cvt->len_cvt;
float *dst = (float *) cvt->buf + cvt->len_cvt * 3;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
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lf = SDL_SwapFloatBE(src[0]);
rf = SDL_SwapFloatBE(src[1]);
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ce = (lf * 0.5f) + (rf * 0.5f);
dst[0] = src[0];
dst[1] = src[1];
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dst[2] = SDL_SwapFloatBE(lf - ce);
dst[3] = SDL_SwapFloatBE(rf - ce);
dst[4] = dst[5] = SDL_SwapFloatBE(ce);
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589
}
} else {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
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591
lf = SDL_SwapFloatLE(src[0]);
rf = SDL_SwapFloatLE(src[1]);
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ce = (lf * 0.5f) + (rf * 0.5f);
dst[0] = src[0];
dst[1] = src[1];
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dst[2] = SDL_SwapFloatLE(lf - ce);
dst[3] = SDL_SwapFloatLE(rf - ce);
dst[4] = dst[5] = SDL_SwapFloatLE(ce);
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602
}
}
}
break;
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}
cvt->len_cvt *= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
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611
}
/* Duplicate a stereo channel to a pseudo-4.0 stream */
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static void SDLCALL
SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
614
{
615
int i;
616
617
#ifdef DEBUG_CONVERT
618
fprintf(stderr, "Converting stereo to quad\n");
619
#endif
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switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
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665
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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;
670
if (SDL_AUDIO_ISBIGENDIAN(format)) {
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688
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690
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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;
720
if (SDL_AUDIO_ISBIGENDIAN(format)) {
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737
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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;
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764
765
766
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;
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768
769
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771
772
773
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784
785
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790
791
792
793
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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;
795
796
797
798
799
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
800
801
}
802
803
804
/* Convert rate up by multiple of 2 */
static void SDLCALL
SDL_RateMUL2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
805
{
806
int i;
807
808
#ifdef DEBUG_CONVERT
809
fprintf(stderr, "Converting audio rate * 2 (mono)\n");
810
811
#endif
812
#define mul2_mono(type) { \
813
814
815
816
817
818
819
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--; \
dst[-1] = dst[-2] = src[0]; \
dst -= 2; \
} \
820
}
821
822
switch (SDL_AUDIO_BITSIZE(format)) {
823
case 8:
824
mul2_mono(Uint8);
825
826
break;
case 16:
827
828
829
830
mul2_mono(Uint16);
break;
case 32:
mul2_mono(Uint32);
831
832
break;
}
833
834
#undef mul2_mono
835
836
837
838
839
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
840
841
}
842
843
/* Convert rate up by multiple of 2, for stereo */
844
845
static void SDLCALL
SDL_RateMUL2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
846
{
847
int i;
848
849
#ifdef DEBUG_CONVERT
850
fprintf(stderr, "Converting audio rate * 2 (stereo)\n");
851
#endif
852
853
#define mul2_stereo(type) { \
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
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) * 2); i; --i) { \
const type r = src[-1]; \
const type l = src[-2]; \
src -= 2; \
dst[-1] = r; \
dst[-2] = l; \
dst[-3] = r; \
dst[-4] = l; \
dst -= 4; \
} \
}
switch (SDL_AUDIO_BITSIZE(format)) {
869
case 8:
870
mul2_stereo(Uint8);
871
872
break;
case 16:
873
874
875
876
mul2_stereo(Uint16);
break;
case 32:
mul2_stereo(Uint32);
877
878
break;
}
879
880
#undef mul2_stereo
881
882
883
884
885
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
886
887
888
}
/* Convert rate up by multiple of 2, for quad */
889
890
static void SDLCALL
SDL_RateMUL2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
891
{
892
int i;
893
894
#ifdef DEBUG_CONVERT
895
fprintf(stderr, "Converting audio rate * 2 (quad)\n");
896
#endif
897
898
#define mul2_quad(type) { \
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
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) * 4); i; --i) { \
const type c1 = src[-1]; \
const type c2 = src[-2]; \
const type c3 = src[-3]; \
const type c4 = src[-4]; \
src -= 4; \
dst[-1] = c1; \
dst[-2] = c2; \
dst[-3] = c3; \
dst[-4] = c4; \
dst[-5] = c1; \
dst[-6] = c2; \
dst[-7] = c3; \
dst[-8] = c4; \
dst -= 8; \
} \
}
switch (SDL_AUDIO_BITSIZE(format)) {
920
case 8:
921
mul2_quad(Uint8);
922
923
break;
case 16:
924
925
926
927
mul2_quad(Uint16);
break;
case 32:
mul2_quad(Uint32);
928
929
break;
}
930
931
#undef mul2_quad
932
933
934
935
936
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
937
938
939
940
}
/* Convert rate up by multiple of 2, for 5.1 */
941
942
static void SDLCALL
SDL_RateMUL2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
943
{
944
int i;
945
946
#ifdef DEBUG_CONVERT
947
fprintf(stderr, "Converting audio rate * 2 (six channels)\n");
948
#endif
949
950
#define mul2_chansix(type) { \
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
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) * 6); i; --i) { \
const type c1 = src[-1]; \
const type c2 = src[-2]; \
const type c3 = src[-3]; \
const type c4 = src[-4]; \
const type c5 = src[-5]; \
const type c6 = src[-6]; \
src -= 6; \
dst[-1] = c1; \
dst[-2] = c2; \
dst[-3] = c3; \
dst[-4] = c4; \
dst[-5] = c5; \
dst[-6] = c6; \
dst[-7] = c1; \
dst[-8] = c2; \
dst[-9] = c3; \
dst[-10] = c4; \
dst[-11] = c5; \
dst[-12] = c6; \
dst -= 12; \
} \
}
switch (SDL_AUDIO_BITSIZE(format)) {
978
case 8:
979
mul2_chansix(Uint8);
980
981
break;
case 16:
982
983
984
985
mul2_chansix(Uint16);
break;
case 32:
mul2_chansix(Uint32);
986
987
break;
}
988
989
#undef mul2_chansix
990
991
992
993
994
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
995
996
}
997
/* Convert rate down by multiple of 2 */
998
999
static void SDLCALL
SDL_RateDIV2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
1000
{