This repository has been archived by the owner on Feb 11, 2021. It is now read-only.
SDL_audiocvt.c
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/*
SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2009 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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/* 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 */
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/* !!! FIXME */
#ifndef assert
#define assert(x)
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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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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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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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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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/* Discard top 4 channels */
static void SDLCALL
SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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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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#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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#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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/* 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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#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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/* 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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#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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/* 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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#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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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting stereo to surround\n");
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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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}
} else {
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
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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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}
}
}
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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}
/* 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)
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting stereo to quad\n");
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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 * 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;
644
if (SDL_AUDIO_ISBIGENDIAN(format)) {
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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;
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if (SDL_AUDIO_ISBIGENDIAN(format)) {
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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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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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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;
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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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int
SDL_ConvertAudio(SDL_AudioCVT * cvt)
779
{
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/* !!! FIXME: (cvt) should be const; stack-copy it here. */
/* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
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/* 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);
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}
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808
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.
*/
809
return NULL; /* no specialized converter code available. */
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}
/*
* 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) {
840
return -1; /* Still no matching converter?! */
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}
}
/* 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);
}
854
return 1; /* added a converter. */
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return 0; /* no conversion necessary. */
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static SDL_AudioFilter
SDL_HandTunedResampleCVT(SDL_AudioCVT * cvt, int dst_channels,
int src_rate, int dst_rate)
865
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/*
* Fill in any future conversions that are specialized to a
* processor, platform, compiler, or library here.
870
return NULL; /* no specialized converter code available. */
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874
static int
SDL_FindFrequencyMultiple(const int src_rate, const int dst_rate)
876
int retval = 0;
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879
880
881
/* If we only built with the arbitrary resamplers, ignore multiples. */
#if !LESS_RESAMPLERS
int lo, hi;
int div;
883
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885
assert(src_rate != 0);
assert(dst_rate != 0);
assert(src_rate != dst_rate);
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889
if (src_rate < dst_rate) {
lo = src_rate;
hi = dst_rate;
891
892
lo = dst_rate;
hi = src_rate;
895
896
/* zero means "not a supported multiple" ... we only do 2x and 4x. */
if ((hi % lo) != 0)
897
return 0; /* not a multiple. */
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div = hi / lo;
retval = ((div == 2) || (div == 4)) ? div : 0;
#endif
903
return retval;
906
907
908
static int
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, int dst_channels,
int src_rate, int dst_rate)
910
911
912
if (src_rate != dst_rate) {
SDL_AudioFilter filter = SDL_HandTunedResampleCVT(cvt, dst_channels,
src_rate, dst_rate);
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917
/* No hand-tuned converter? Try the autogenerated ones. */
if (filter == NULL) {
int i;
const int upsample = (src_rate < dst_rate) ? 1 : 0;
918
919
const int multiple =
SDL_FindFrequencyMultiple(src_rate, dst_rate);
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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;
}
}
932
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934
935
if (filter == NULL) {
return -1; /* Still no matching converter?! */
}
}
937
938
939
940
/* 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);
941
cvt->len_mult *= (int) SDL_ceil(mult);
942
943
944
945
cvt->len_ratio *= mult;
} else {
cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
}
947
return 1; /* added a converter. */
950
return 0; /* no conversion necessary. */
952
953
954
955
956
/* 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.
957
*/
958
959
960
int
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
961
962
SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
963
{
964
965
966
967
968
969
970
971
/*
* !!! 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.
*/
972
973
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978
/* there are no unsigned types over 16 bits, so catch this upfront. */
if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
return -1;
}
if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
return -1;
}
979
980
981
982
983
/* prevent possible divisions by zero, etc. */
if ((src_rate == 0) || (dst_rate == 0)) {
return -1;
}
984
#ifdef DEBUG_CONVERT
985
printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
986
987
src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
#endif
989
/* Start off with no conversion necessary */
990
SDL_zerop(cvt);
991
992
cvt->src_format = src_fmt;
cvt->dst_format = dst_fmt;
993
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995
996
997
cvt->needed = 0;
cvt->filter_index = 0;
cvt->filters[0] = NULL;
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
998
cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
1000
/* Convert data types, if necessary. Updates (cvt). */