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SDL_audiocvt.c
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/*
Simple DirectMedia Layer
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Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
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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 */
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#include "SDL.h"
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#include "SDL_audio.h"
#include "SDL_audio_c.h"
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#include "SDL_loadso.h"
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#include "SDL_assert.h"
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#include "../SDL_dataqueue.h"
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#include "SDL_cpuinfo.h"
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#ifdef __SSE3__
#define HAVE_SSE3_INTRINSICS 1
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#endif
#if HAVE_SSE3_INTRINSICS
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/* Convert from stereo to mono. Average left and right. */
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static void SDLCALL
SDL_ConvertStereoToMono_SSE3(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float *dst = (float *) cvt->buf;
const float *src = dst;
int i = cvt->len_cvt / 8;
LOG_DEBUG_CONVERT("stereo", "mono (using SSE3)");
SDL_assert(format == AUDIO_F32SYS);
/* We can only do this if dst is aligned to 16 bytes; since src is the
same pointer and it moves by 2, it can't be forcibly aligned. */
if ((((size_t) dst) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 divby2 = _mm_set1_ps(0.5f);
while (i >= 4) { /* 4 * float32 */
_mm_store_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_load_ps(src), _mm_load_ps(src+4)), divby2));
i -= 4; src += 8; dst += 4;
}
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (src[0] + src[1]) * 0.5f;
dst++; i--; src += 2;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
#endif
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/* Convert from stereo to mono. Average left and right. */
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static void SDLCALL
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SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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float *dst = (float *) cvt->buf;
const float *src = dst;
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int i;
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LOG_DEBUG_CONVERT("stereo", "mono");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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*(dst++) = (src[0] + src[1]) * 0.5f;
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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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/* Convert from 5.1 to stereo. Average left and right, distribute center, discard LFE. */
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static void SDLCALL
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SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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float *dst = (float *) cvt->buf;
const float *src = dst;
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int i;
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LOG_DEBUG_CONVERT("5.1", "stereo");
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SDL_assert(format == AUDIO_F32SYS);
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/* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
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for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
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const float front_center_distributed = src[2] * 0.5f;
dst[0] = (src[0] + front_center_distributed + src[4]) / 2.5f; /* left */
dst[1] = (src[1] + front_center_distributed + src[5]) / 2.5f; /* right */
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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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/* Convert from quad to stereo. Average left and right. */
static void SDLCALL
SDL_ConvertQuadToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float *dst = (float *) cvt->buf;
const float *src = dst;
int i;
LOG_DEBUG_CONVERT("quad", "stereo");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / (sizeof (float) * 4); i; --i, src += 4, dst += 2) {
dst[0] = (src[0] + src[2]) * 0.5f; /* left */
dst[1] = (src[1] + src[3]) * 0.5f; /* right */
}
cvt->len_cvt /= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Convert from 7.1 to 5.1. Distribute sides across front and back. */
static void SDLCALL
SDL_Convert71To51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float *dst = (float *) cvt->buf;
const float *src = dst;
int i;
LOG_DEBUG_CONVERT("7.1", "5.1");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / (sizeof (float) * 8); i; --i, src += 8, dst += 6) {
const float surround_left_distributed = src[6] * 0.5f;
const float surround_right_distributed = src[7] * 0.5f;
dst[0] = (src[0] + surround_left_distributed) / 1.5f; /* FL */
dst[1] = (src[1] + surround_right_distributed) / 1.5f; /* FR */
dst[2] = src[2] / 1.5f; /* CC */
dst[3] = src[3] / 1.5f; /* LFE */
dst[4] = (src[4] + surround_left_distributed) / 1.5f; /* BL */
dst[5] = (src[5] + surround_right_distributed) / 1.5f; /* BR */
}
cvt->len_cvt /= 8;
cvt->len_cvt *= 6;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Convert from 5.1 to quad. Distribute center across front, discard LFE. */
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static void SDLCALL
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SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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float *dst = (float *) cvt->buf;
const float *src = dst;
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int i;
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LOG_DEBUG_CONVERT("5.1", "quad");
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SDL_assert(format == AUDIO_F32SYS);
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/* SDL's 4.0 layout: FL+FR+BL+BR */
/* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
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for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
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const float front_center_distributed = src[2] * 0.5f;
dst[0] = (src[0] + front_center_distributed) / 1.5f; /* FL */
dst[1] = (src[1] + front_center_distributed) / 1.5f; /* FR */
dst[2] = src[4] / 1.5f; /* BL */
dst[3] = src[5] / 1.5f; /* BR */
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}
cvt->len_cvt /= 6;
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
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/* Upmix mono to stereo (by duplication) */
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static void SDLCALL
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SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
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int i;
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LOG_DEBUG_CONVERT("mono", "stereo");
SDL_assert(format == AUDIO_F32SYS);
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for (i = cvt->len_cvt / sizeof (float); i; --i) {
src--;
dst -= 2;
dst[0] = dst[1] = *src;
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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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/* Upmix stereo to a pseudo-5.1 stream */
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static void SDLCALL
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SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
int i;
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float lf, rf, ce;
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);
LOG_DEBUG_CONVERT("stereo", "5.1");
SDL_assert(format == AUDIO_F32SYS);
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for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
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dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
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ce = (lf + rf) * 0.5f;
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/* !!! FIXME: FL and FR may clip */
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dst[0] = lf + (lf - ce); /* FL */
dst[1] = rf + (rf - ce); /* FR */
dst[2] = ce; /* FC */
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dst[3] = 0; /* LFE (only meant for special LFE effects) */
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dst[4] = lf; /* BL */
dst[5] = rf; /* BR */
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}
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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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/* Upmix quad to a pseudo-5.1 stream */
static void SDLCALL
SDL_ConvertQuadTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
float lf, rf, lb, rb, ce;
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 3 / 2);
LOG_DEBUG_CONVERT("quad", "5.1");
SDL_assert(format == AUDIO_F32SYS);
SDL_assert(cvt->len_cvt % (sizeof(float) * 4) == 0);
for (i = cvt->len_cvt / (sizeof(float) * 4); i; --i) {
dst -= 6;
src -= 4;
lf = src[0];
rf = src[1];
lb = src[2];
rb = src[3];
ce = (lf + rf) * 0.5f;
/* !!! FIXME: FL and FR may clip */
dst[0] = lf + (lf - ce); /* FL */
dst[1] = rf + (rf - ce); /* FR */
dst[2] = ce; /* FC */
dst[3] = 0; /* LFE (only meant for special LFE effects) */
dst[4] = lb; /* BL */
dst[5] = rb; /* BR */
}
cvt->len_cvt = cvt->len_cvt * 3 / 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Upmix stereo to a pseudo-4.0 stream (by duplication) */
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static void SDLCALL
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SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
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float lf, rf;
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int i;
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LOG_DEBUG_CONVERT("stereo", "quad");
SDL_assert(format == AUDIO_F32SYS);
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for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
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dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
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dst[0] = lf; /* FL */
dst[1] = rf; /* FR */
dst[2] = lf; /* BL */
dst[3] = rf; /* BR */
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}
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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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/* Upmix 5.1 to 7.1 */
static void SDLCALL
SDL_Convert51To71(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float lf, rf, lb, rb, ls, rs;
int i;
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 4 / 3);
LOG_DEBUG_CONVERT("5.1", "7.1");
SDL_assert(format == AUDIO_F32SYS);
SDL_assert(cvt->len_cvt % (sizeof(float) * 6) == 0);
for (i = cvt->len_cvt / (sizeof(float) * 6); i; --i) {
dst -= 8;
src -= 6;
lf = src[0];
rf = src[1];
lb = src[4];
rb = src[5];
ls = (lf + lb) * 0.5f;
rs = (rf + rb) * 0.5f;
/* !!! FIXME: these four may clip */
lf += lf - ls;
rf += rf - ls;
lb += lb - ls;
rb += rb - ls;
dst[3] = src[3]; /* LFE */
dst[2] = src[2]; /* FC */
dst[7] = rs; /* SR */
dst[6] = ls; /* SL */
dst[5] = rb; /* BR */
dst[4] = lb; /* BL */
dst[1] = rf; /* FR */
dst[0] = lf; /* FL */
}
cvt->len_cvt = cvt->len_cvt * 4 / 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
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static int
SDL_ResampleAudioSimple(const int chans, const double rate_incr,
float *last_sample, const float *inbuf,
const int inbuflen, float *outbuf, const int outbuflen)
{
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const int framelen = chans * (int)sizeof (float);
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const int total = (inbuflen / framelen);
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const int finalpos = (total * chans) - chans;
const int dest_samples = (int)(((double)total) * rate_incr);
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const double src_incr = 1.0 / rate_incr;
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float *dst;
double idx;
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int i;
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SDL_assert((dest_samples * framelen) <= outbuflen);
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SDL_assert((inbuflen % framelen) == 0);
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if (rate_incr > 1.0) { /* upsample */
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float *target = (outbuf + chans);
dst = outbuf + (dest_samples * chans);
idx = (double) total;
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if (chans == 1) {
const float final_sample = inbuf[finalpos];
float earlier_sample = inbuf[finalpos];
while (dst > target) {
const int pos = ((int) idx) * chans;
const float *src = &inbuf[pos];
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const float val = *(--src);
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SDL_assert(pos >= 0.0);
*(--dst) = (val + earlier_sample) * 0.5f;
earlier_sample = val;
idx -= src_incr;
}
/* do last sample, interpolated against previous run's state. */
*(--dst) = (inbuf[0] + last_sample[0]) * 0.5f;
*last_sample = final_sample;
} else if (chans == 2) {
const float final_sample2 = inbuf[finalpos+1];
const float final_sample1 = inbuf[finalpos];
float earlier_sample2 = inbuf[finalpos];
float earlier_sample1 = inbuf[finalpos-1];
while (dst > target) {
const int pos = ((int) idx) * chans;
const float *src = &inbuf[pos];
const float val2 = *(--src);
const float val1 = *(--src);
SDL_assert(pos >= 0.0);
*(--dst) = (val2 + earlier_sample2) * 0.5f;
*(--dst) = (val1 + earlier_sample1) * 0.5f;
earlier_sample2 = val2;
earlier_sample1 = val1;
idx -= src_incr;
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}
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/* do last sample, interpolated against previous run's state. */
*(--dst) = (inbuf[1] + last_sample[1]) * 0.5f;
*(--dst) = (inbuf[0] + last_sample[0]) * 0.5f;
last_sample[1] = final_sample2;
last_sample[0] = final_sample1;
} else {
const float *earlier_sample = &inbuf[finalpos];
float final_sample[8];
SDL_memcpy(final_sample, &inbuf[finalpos], framelen);
while (dst > target) {
const int pos = ((int) idx) * chans;
const float *src = &inbuf[pos];
SDL_assert(pos >= 0.0);
for (i = chans - 1; i >= 0; i--) {
const float val = *(--src);
*(--dst) = (val + earlier_sample[i]) * 0.5f;
}
earlier_sample = src;
idx -= src_incr;
}
/* do last sample, interpolated against previous run's state. */
for (i = chans - 1; i >= 0; i--) {
const float val = inbuf[i];
*(--dst) = (val + last_sample[i]) * 0.5f;
}
SDL_memcpy(last_sample, final_sample, framelen);
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}
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dst = (outbuf + (dest_samples * chans));
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} else { /* downsample */
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float *target = (outbuf + (dest_samples * chans));
dst = outbuf;
idx = 0.0;
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if (chans == 1) {
float last = *last_sample;
while (dst < target) {
const int pos = ((int) idx) * chans;
const float val = inbuf[pos];
SDL_assert(pos <= finalpos);
*(dst++) = (val + last) * 0.5f;
last = val;
idx += src_incr;
}
*last_sample = last;
} else if (chans == 2) {
float last1 = last_sample[0];
float last2 = last_sample[1];
while (dst < target) {
const int pos = ((int) idx) * chans;
const float val1 = inbuf[pos];
const float val2 = inbuf[pos+1];
SDL_assert(pos <= finalpos);
*(dst++) = (val1 + last1) * 0.5f;
*(dst++) = (val2 + last2) * 0.5f;
last1 = val1;
last2 = val2;
idx += src_incr;
}
last_sample[0] = last1;
last_sample[1] = last2;
} else {
while (dst < target) {
const int pos = ((int) idx) * chans;
const float *src = &inbuf[pos];
SDL_assert(pos <= finalpos);
for (i = 0; i < chans; i++) {
const float val = *(src++);
*(dst++) = (val + last_sample[i]) * 0.5f;
last_sample[i] = val;
}
idx += src_incr;
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}
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}
}
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return (int) ((dst - outbuf) * ((int) sizeof (float)));
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}
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/* We keep one special-case fast path around for an extremely common audio format. */
static int
SDL_ResampleAudioSimple_si16_c2(const double rate_incr,
Sint16 *last_sample, const Sint16 *inbuf,
const int inbuflen, Sint16 *outbuf, const int outbuflen)
{
const int chans = 2;
const int framelen = 4; /* stereo 16 bit */
const int total = (inbuflen / framelen);
const int finalpos = (total * chans) - chans;
const int dest_samples = (int)(((double)total) * rate_incr);
const double src_incr = 1.0 / rate_incr;
Sint16 *dst;
double idx;
SDL_assert((dest_samples * framelen) <= outbuflen);
SDL_assert((inbuflen % framelen) == 0);
if (rate_incr > 1.0) {
Sint16 *target = (outbuf + chans);
const Sint16 final_right = inbuf[finalpos+1];
const Sint16 final_left = inbuf[finalpos];
Sint16 earlier_right = inbuf[finalpos-1];
Sint16 earlier_left = inbuf[finalpos-2];
dst = outbuf + (dest_samples * chans);
idx = (double) total;
while (dst > target) {
const int pos = ((int) idx) * chans;
const Sint16 *src = &inbuf[pos];
const Sint16 right = *(--src);
const Sint16 left = *(--src);
SDL_assert(pos >= 0.0);
*(--dst) = (((Sint32) right) + ((Sint32) earlier_right)) >> 1;
*(--dst) = (((Sint32) left) + ((Sint32) earlier_left)) >> 1;
earlier_right = right;
earlier_left = left;
idx -= src_incr;
}
/* do last sample, interpolated against previous run's state. */
*(--dst) = (((Sint32) inbuf[1]) + ((Sint32) last_sample[1])) >> 1;
*(--dst) = (((Sint32) inbuf[0]) + ((Sint32) last_sample[0])) >> 1;
last_sample[1] = final_right;
last_sample[0] = final_left;
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dst = (outbuf + (dest_samples * chans));
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} else {
Sint16 *target = (outbuf + (dest_samples * chans));
dst = outbuf;
idx = 0.0;
while (dst < target) {
const int pos = ((int) idx) * chans;
const Sint16 *src = &inbuf[pos];
const Sint16 left = *(src++);
const Sint16 right = *(src++);
SDL_assert(pos <= finalpos);
*(dst++) = (((Sint32) left) + ((Sint32) last_sample[0])) >> 1;
*(dst++) = (((Sint32) right) + ((Sint32) last_sample[1])) >> 1;
last_sample[0] = left;
last_sample[1] = right;
idx += src_incr;
}
}
return (int) ((dst - outbuf) * ((int) sizeof (Sint16)));
}
static void SDLCALL
SDL_ResampleCVT_si16_c2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint16 *src = (const Sint16 *) cvt->buf;
const int srclen = cvt->len_cvt;
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Sint16 *dst = (Sint16 *) cvt->buf;
const int dstlen = (cvt->len * cvt->len_mult);
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Sint16 state[2];
state[0] = src[0];
state[1] = src[1];
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SDL_assert(format == AUDIO_S16SYS);
cvt->len_cvt = SDL_ResampleAudioSimple_si16_c2(cvt->rate_incr, state, src, srclen, dst, dstlen);
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
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601
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) {
602
return SDL_SetError("No buffer allocated for conversion");
603
}
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/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if (cvt->filters[0] == NULL) {
608
return 0;
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611
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613
}
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0] (cvt, cvt->src_format);
614
return 0;
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616
}
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static void SDLCALL
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
619
{
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622
#if DEBUG_CONVERT
printf("Converting byte order\n");
#endif
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switch (SDL_AUDIO_BITSIZE(format)) {
#define CASESWAP(b) \
case b: { \
Uint##b *ptr = (Uint##b *) cvt->buf; \
int i; \
for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
*ptr = SDL_Swap##b(*ptr); \
} \
break; \
}
CASESWAP(16);
CASESWAP(32);
CASESWAP(64);
#undef CASESWAP
default: SDL_assert(!"unhandled byteswap datatype!"); break;
}
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if (cvt->filters[++cvt->filter_index]) {
/* flip endian flag for data. */
if (format & SDL_AUDIO_MASK_ENDIAN) {
format &= ~SDL_AUDIO_MASK_ENDIAN;
} else {
format |= SDL_AUDIO_MASK_ENDIAN;
}
cvt->filters[cvt->filter_index](cvt, format);
}
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}
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static int
SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter)
{
if (cvt->filter_index >= SDL_AUDIOCVT_MAX_FILTERS) {
return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS);
}
if (filter == NULL) {
return SDL_SetError("Audio filter pointer is NULL");
}
cvt->filters[cvt->filter_index++] = filter;
cvt->filters[cvt->filter_index] = NULL; /* Moving terminator */
return 0;
}
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static int
670
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
671
{
672
int retval = 0; /* 0 == no conversion necessary. */
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if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
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if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
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retval = 1; /* added a converter. */
}
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if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
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const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
const Uint16 dst_bitsize = 32;
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SDL_AudioFilter filter = NULL;
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switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
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case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
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case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
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}
695
if (!filter) {
696
return SDL_SetError("No conversion from source format to float available");
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}
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if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
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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);
}
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retval = 1; /* added a converter. */
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}
713
return retval;
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}
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static int
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
718
{
719
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721
int retval = 0; /* 0 == no conversion necessary. */
if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
722
723
const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
const Uint16 src_bitsize = 32;
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SDL_AudioFilter filter = NULL;
switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
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case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
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case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
if (!filter) {
735
return SDL_SetError("No conversion from float to destination format available");
736
}
737
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739
740
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
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749
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751
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);
}
retval = 1; /* added a converter. */
}
if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
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754
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
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retval = 1; /* added a converter. */
}
return retval;
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760
}
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765
static void
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
const int srclen = cvt->len_cvt;
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767
float *dst = (float *) cvt->buf;
const int dstlen = (cvt->len * cvt->len_mult);
768
float state[8];
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771
SDL_assert(format == AUDIO_F32SYS);
772
SDL_memcpy(state, src, chans*sizeof(*src));
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774
cvt->len_cvt = SDL_ResampleAudioSimple(chans, cvt->rate_incr, state, src, srclen, dst, dstlen);
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if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
!!! FIXME: store channel info, so we have to have function entry
!!! FIXME: points for each supported channel count and multiple
!!! FIXME: vs arbitrary. When we rev the ABI, clean this up. */
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785
#define RESAMPLER_FUNCS(chans) \
static void SDLCALL \
786
787
SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
SDL_ResampleCVT(cvt, chans, format); \
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789
790
791
792
793
794
795
}
RESAMPLER_FUNCS(1)
RESAMPLER_FUNCS(2)
RESAMPLER_FUNCS(4)
RESAMPLER_FUNCS(6)
RESAMPLER_FUNCS(8)
#undef RESAMPLER_FUNCS
796
static SDL_AudioFilter
797
ChooseCVTResampler(const int dst_channels)
798
{
799
800
801
802
803
804
805
switch (dst_channels) {
case 1: return SDL_ResampleCVT_c1;
case 2: return SDL_ResampleCVT_c2;
case 4: return SDL_ResampleCVT_c4;
case 6: return SDL_ResampleCVT_c6;
case 8: return SDL_ResampleCVT_c8;
default: break;
806
807
}
808
return NULL;
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810
811
812
813
814
815
816
817
818
819
820
}
static int
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
const int src_rate, const int dst_rate)
{
SDL_AudioFilter filter;
if (src_rate == dst_rate) {
return 0; /* no conversion necessary. */
}
821
filter = ChooseCVTResampler(dst_channels);
822
823
824
if (filter == NULL) {
return SDL_SetError("No conversion available for these rates");
}
825
826
/* Update (cvt) with filter details... */
827
828
829
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
830
831
832
833
834
835
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);
836
837
}
838
return 1; /* added a converter. */
839
840
}
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
static SDL_bool
SDL_SupportedAudioFormat(const SDL_AudioFormat fmt)
{
switch (fmt) {
case AUDIO_U8:
case AUDIO_S8:
case AUDIO_U16LSB:
case AUDIO_S16LSB:
case AUDIO_U16MSB:
case AUDIO_S16MSB:
case AUDIO_S32LSB:
case AUDIO_S32MSB:
case AUDIO_F32LSB:
case AUDIO_F32MSB:
return SDL_TRUE; /* supported. */
default:
break;
}
return SDL_FALSE; /* unsupported. */
}
static SDL_bool
SDL_SupportedChannelCount(const int channels)
{
switch (channels) {
case 1: /* mono */
case 2: /* stereo */
case 4: /* quad */
case 6: /* 5.1 */
872
873
case 8: /* 7.1 */
return SDL_TRUE; /* supported. */
874
875
876
877
878
879
880
881
default:
break;
}
return SDL_FALSE; /* unsupported. */
}
882
883
/* Creates a set of audio filters to convert from one format to another.
884
885
Returns 0 if no conversion is needed, 1 if the audio filter is set up,
or -1 if an error like invalid parameter, unsupported format, etc. occurred.
886
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888
889
890
891
892
893
894
895
896
897
*/
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)
{
/* Sanity check target pointer */
if (cvt == NULL) {
return SDL_InvalidParamError("cvt");
}
898
899
900
/* Make sure we zero out the audio conversion before error checking */
SDL_zerop(cvt);
901
if (!SDL_SupportedAudioFormat(src_fmt)) {
902
return SDL_SetError("Invalid source format");
903
} else if (!SDL_SupportedAudioFormat(dst_fmt)) {
904
return SDL_SetError("Invalid destination format");
905
906
907
908
909
910
911
912
} else if (!SDL_SupportedChannelCount(src_channels)) {
return SDL_SetError("Invalid source channels");
} else if (!SDL_SupportedChannelCount(dst_channels)) {
return SDL_SetError("Invalid destination channels");
} else if (src_rate == 0) {
return SDL_SetError("Source rate is zero");
} else if (dst_rate == 0) {
return SDL_SetError("Destination rate is zero");
913
914
}
915
#if DEBUG_CONVERT
916
917
918
919
920
921
922
923
924
925
926
927
928
929
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 */
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);
930
931
932
/* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
SDL_ChooseAudioConverters();
933
934
935
936
937
938
939
940
941
942
943
944
945
/* SDL now favors float32 as its preferred internal format, and considers
everything else to be a degenerate case that we might have to make
multiple passes over the data to convert to and from float32 as
necessary. That being said, we keep one special case around for
efficiency: stereo data in Sint16 format, in the native byte order,
that only needs resampling. This is likely to be the most popular
legacy format, that apps, hardware and the OS are likely to be able
to process directly, so we handle this one case directly without
unnecessary conversions. This means that apps on embedded devices
without floating point hardware should consider aiming for this
format as well. */
if ((src_channels == 2) && (dst_channels == 2) && (src_fmt == AUDIO_S16SYS) && (dst_fmt == AUDIO_S16SYS) && (src_rate != dst_rate)) {
cvt->needed = 1;
946
947
948
if (SDL_AddAudioCVTFilter(cvt, SDL_ResampleCVT_si16_c2) < 0) {
return -1;
}
949
950
951
952
953
954
955
956
957
958
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;
}
959
960
961
962
963
964
965
966
967
/* Type conversion goes like this now:
- byteswap to CPU native format first if necessary.
- convert to native Float32 if necessary.
- resample and change channel count if necessary.
- convert back to native format.
- byteswap back to foreign format if necessary.
The expectation is we can process data faster in float32
(possibly with SIMD), and making several passes over the same
968
buffer is likely to be CPU cache-friendly, avoiding the
969
970
971
972
biggest performance hit in modern times. Previously we had
(script-generated) custom converters for every data type and
it was a bloat on SDL compile times and final library size. */
973
974
975
976
977
978
979
980
/* see if we can skip float conversion entirely. */
if (src_rate == dst_rate && src_channels == dst_channels) {
if (src_fmt == dst_fmt) {
return 0;
}
/* just a byteswap needed? */
if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
981
982
983
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
984
985
986
cvt->needed = 1;
return 1;
}
987
988
}
989
/* Convert data types, if necessary. Updates (cvt). */
990
if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
991
992
993
994
return -1; /* shouldn't happen, but just in case... */
}
/* Channel conversion */
995
996
997
if (src_channels < dst_channels) {
/* Upmixing */
/* Mono -> Stereo [-> ...] */
998
if ((src_channels == 1) && (dst_channels > 1)) {
999
1000
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
return -1;