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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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SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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float *dst = (float *) cvt->buf;
const float *src = dst;
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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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SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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float *dst = (float *) cvt->buf;
const float *src = dst;
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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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SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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float *dst = (float *) cvt->buf;
const float *src = dst;
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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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SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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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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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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SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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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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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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SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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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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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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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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/* SDL's resampler uses a "bandlimited interpolation" algorithm:
https://ccrma.stanford.edu/~jos/resample/ */
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#define RESAMPLER_ZERO_CROSSINGS 5
#define RESAMPLER_BITS_PER_SAMPLE 16
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))
#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1)
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/* This is a "modified" bessel function, so you can't use POSIX j0() */
static double
bessel(const double x)
{
const double xdiv2 = x / 2.0;
double i0 = 1.0f;
double f = 1.0f;
int i = 1;
while (SDL_TRUE) {
const double diff = SDL_pow(xdiv2, i * 2) / SDL_pow(f, 2);
if (diff < 1.0e-21f) {
break;
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}
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i0 += diff;
i++;
f *= (double) i;
}
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return i0;
}
/* build kaiser table with cardinal sine applied to it, and array of differences between elements. */
static void
kaiser_and_sinc(float *table, float *diffs, const int tablelen, const double beta)
{
const int lenm1 = tablelen - 1;
const int lenm1div2 = lenm1 / 2;
int i;
table[0] = 1.0f;
for (i = 1; i < tablelen; i++) {
const double kaiser = bessel(beta * SDL_sqrt(1.0 - SDL_pow(((i - lenm1) / 2.0) / lenm1div2, 2.0))) / bessel(beta);
table[tablelen - i] = (float) kaiser;
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}
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for (i = 1; i < tablelen; i++) {
const float x = (((float) i) / ((float) RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) * ((float) M_PI);
table[i] *= SDL_sinf(x) / x;
diffs[i - 1] = table[i] - table[i - 1];
}
diffs[lenm1] = 0.0f;
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}
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static SDL_SpinLock ResampleFilterSpinlock = 0;
static float *ResamplerFilter = NULL;
static float *ResamplerFilterDifference = NULL;
int
SDL_PrepareResampleFilter(void)
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{
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SDL_AtomicLock(&ResampleFilterSpinlock);
if (!ResamplerFilter) {
/* if dB > 50, beta=(0.1102 * (dB - 8.7)), according to Matlab. */
const double dB = 80.0;
const double beta = 0.1102 * (dB - 8.7);
const size_t alloclen = RESAMPLER_FILTER_SIZE * sizeof (float);
ResamplerFilter = (float *) SDL_malloc(alloclen);
if (!ResamplerFilter) {
SDL_AtomicUnlock(&ResampleFilterSpinlock);
return SDL_OutOfMemory();
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}
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ResamplerFilterDifference = (float *) SDL_malloc(alloclen);
if (!ResamplerFilterDifference) {
SDL_free(ResamplerFilter);
ResamplerFilter = NULL;
SDL_AtomicUnlock(&ResampleFilterSpinlock);
return SDL_OutOfMemory();
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}
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kaiser_and_sinc(ResamplerFilter, ResamplerFilterDifference, RESAMPLER_FILTER_SIZE, beta);
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}
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SDL_AtomicUnlock(&ResampleFilterSpinlock);
return 0;
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}
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void
SDL_FreeResampleFilter(void)
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{
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SDL_free(ResamplerFilter);
SDL_free(ResamplerFilterDifference);
ResamplerFilter = NULL;
ResamplerFilterDifference = NULL;
}
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static int
ResamplerPadding(const int inrate, const int outrate)
{
return (inrate > outrate) ? (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate))) : RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
}
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/* lpadding and rpadding are expected to be buffers of (ResamplePadding(inrate, outrate) * chans * sizeof (float)) bytes. */
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static int
SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
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float *lpadding, float *rpadding, const float *inbuf,
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const int inbuflen, float *outbuf, const int outbuflen)
{
const float outtimeincr = 1.0f / ((float) outrate);
const float ratio = ((float) outrate) / ((float) inrate);
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const int paddinglen = ResamplerPadding(inrate, outrate);
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const int framelen = chans * (int)sizeof (float);
const int inframes = inbuflen / framelen;
const int wantedoutframes = (int) ((inbuflen / framelen) * ratio); /* outbuflen isn't total to write, it's total available. */
const int maxoutframes = outbuflen / framelen;
const int outframes = (wantedoutframes < maxoutframes) ? wantedoutframes : maxoutframes;
float *dst = outbuf;
float outtime = 0.0f;
int i, j, chan;
for (i = 0; i < outframes; i++) {
const int srcindex = (int) (outtime * inrate);
const float finrate = (float) inrate;
const float intime = ((float) srcindex) / finrate;
const float innexttime = ((float) (srcindex + 1)) / finrate;
const float interpolation1 = 1.0f - (innexttime - outtime) / (innexttime - intime);
const int filterindex1 = (int) (interpolation1 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
const float interpolation2 = 1.0f - interpolation1;
const int filterindex2 = interpolation2 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
for (chan = 0; chan < chans; chan++) {
float outsample = 0.0f;
/* do this twice to calculate the sample, once for the "left wing" and then same for the right. */
/* !!! FIXME: do both wings in one loop */
for (j = 0; (filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
const int srcframe = srcindex - j;
Sep 22, 2017
Sep 22, 2017
509
510
/* !!! FIXME: we can bubble this conditional out of here by doing a pre loop. */
const float insample = (srcframe < 0) ? lpadding[((paddinglen + srcframe) * chans) + chan] : inbuf[(srcframe * chans) + chan];
Sep 21, 2017
Sep 21, 2017
511
512
outsample += (insample * (ResamplerFilter[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation1 * ResamplerFilterDifference[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
}
Jan 23, 2017
Jan 23, 2017
513
Sep 21, 2017
Sep 21, 2017
514
515
for (j = 0; (filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
const int srcframe = srcindex + 1 + j;
Sep 22, 2017
Sep 22, 2017
516
517
/* !!! FIXME: we can bubble this conditional out of here by doing a post loop. */
const float insample = (srcframe >= inframes) ? rpadding[((srcframe - inframes) * chans) + chan] : inbuf[(srcframe * chans) + chan];
Sep 21, 2017
Sep 21, 2017
518
519
520
521
522
523
outsample += (insample * (ResamplerFilter[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation2 * ResamplerFilterDifference[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
}
*(dst++) = outsample;
}
outtime += outtimeincr;
Jan 23, 2017
Jan 23, 2017
524
525
}
Sep 21, 2017
Sep 21, 2017
526
527
return outframes * chans * sizeof (float);
}
528
529
530
531
532
533
534
535
536
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) {
Nov 5, 2016
Nov 5, 2016
537
return SDL_SetError("No buffer allocated for conversion");
Nov 5, 2016
Nov 5, 2016
539
540
541
542
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if (cvt->filters[0] == NULL) {
Nov 5, 2016
Nov 5, 2016
543
return 0;
544
545
546
547
548
}
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0] (cvt, cvt->src_format);
Nov 5, 2016
Nov 5, 2016
549
return 0;
Nov 5, 2016
Nov 5, 2016
552
553
static void SDLCALL
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Nov 5, 2016
Nov 5, 2016
555
556
557
#if DEBUG_CONVERT
printf("Converting byte order\n");
#endif
Nov 5, 2016
Nov 5, 2016
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
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;
}
Nov 5, 2016
Nov 5, 2016
579
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581
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584
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586
587
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);
}
Jun 12, 2017
Jun 12, 2017
590
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592
593
594
595
596
597
598
599
600
601
602
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;
}
Nov 5, 2016
Nov 5, 2016
605
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
Nov 5, 2016
Nov 5, 2016
607
int retval = 0; /* 0 == no conversion necessary. */
Nov 5, 2016
Nov 5, 2016
609
if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
Jun 12, 2017
Jun 12, 2017
610
611
612
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
Nov 5, 2016
Nov 5, 2016
613
614
retval = 1; /* added a converter. */
}
Nov 5, 2016
Nov 5, 2016
616
if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
Nov 5, 2016
Nov 5, 2016
617
618
const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
const Uint16 dst_bitsize = 32;
Nov 5, 2016
Nov 5, 2016
619
SDL_AudioFilter filter = NULL;
Nov 5, 2016
Nov 5, 2016
620
Nov 5, 2016
Nov 5, 2016
621
622
623
624
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;
Nov 7, 2016
Nov 7, 2016
625
case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
Nov 5, 2016
Nov 5, 2016
626
627
case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
Nov 5, 2016
Nov 5, 2016
630
if (!filter) {
Aug 18, 2017
Aug 18, 2017
631
return SDL_SetError("No conversion from source format to float available");
Nov 5, 2016
Nov 5, 2016
632
633
}
Jun 12, 2017
Jun 12, 2017
634
635
636
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
637
638
639
640
641
642
643
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);
}
Nov 5, 2016
Nov 5, 2016
644
Nov 5, 2016
Nov 5, 2016
645
retval = 1; /* added a converter. */
Nov 5, 2016
Nov 5, 2016
648
return retval;
Nov 5, 2016
Nov 5, 2016
651
652
static int
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
Nov 5, 2016
Nov 5, 2016
654
655
656
int retval = 0; /* 0 == no conversion necessary. */
if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
Nov 5, 2016
Nov 5, 2016
657
658
const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
const Uint16 src_bitsize = 32;
Nov 5, 2016
Nov 5, 2016
659
660
661
662
663
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;
Nov 7, 2016
Nov 7, 2016
664
case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
Nov 5, 2016
Nov 5, 2016
665
666
667
668
669
case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
if (!filter) {
Aug 18, 2017
Aug 18, 2017
670
return SDL_SetError("No conversion from float to destination format available");
Nov 5, 2016
Nov 5, 2016
671
}
Jun 12, 2017
Jun 12, 2017
673
674
675
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
Nov 5, 2016
Nov 5, 2016
676
677
678
679
680
681
682
683
684
685
686
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)) {
Jun 12, 2017
Jun 12, 2017
687
688
689
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
Nov 5, 2016
Nov 5, 2016
690
691
692
693
retval = 1; /* added a converter. */
}
return retval;
Jan 9, 2017
Jan 9, 2017
696
697
698
static void
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
{
Sep 21, 2017
Sep 21, 2017
699
700
701
/* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator).
!!! FIXME in 2.1: We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
!!! FIXME in 2.1: so we steal the ninth and tenth slot. :( */
Sep 22, 2017
Sep 22, 2017
702
703
const int inrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1];
const int outrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS];
Jan 9, 2017
Jan 9, 2017
704
705
const float *src = (const float *) cvt->buf;
const int srclen = cvt->len_cvt;
Sep 21, 2017
Sep 21, 2017
706
707
708
709
710
/*float *dst = (float *) cvt->buf;
const int dstlen = (cvt->len * cvt->len_mult);*/
/* !!! FIXME: remove this if we can get the resampler to work in-place again. */
float *dst = (float *) (cvt->buf + srclen);
const int dstlen = (cvt->len * cvt->len_mult) - srclen;
Sep 22, 2017
Sep 22, 2017
711
const int paddingsamples = (ResamplerPadding(inrate, outrate) * chans);
Sep 22, 2017
Sep 22, 2017
712
float *padding;
Jan 9, 2017
Jan 9, 2017
713
714
715
SDL_assert(format == AUDIO_F32SYS);
Sep 22, 2017
Sep 22, 2017
716
/* we keep no streaming state here, so pad with silence on both ends. */
Sep 22, 2017
Sep 22, 2017
717
718
719
720
721
padding = SDL_stack_alloc(float, paddingsamples);
if (!padding) {
SDL_OutOfMemory();
return;
}
Sep 22, 2017
Sep 22, 2017
722
SDL_memset(padding, '\0', paddingsamples * sizeof (float));
Sep 21, 2017
Sep 21, 2017
723
Sep 22, 2017
Sep 22, 2017
724
cvt->len_cvt = SDL_ResampleAudio(chans, inrate, outrate, padding, padding, src, srclen, dst, dstlen);
Sep 21, 2017
Sep 21, 2017
725
Sep 22, 2017
Sep 22, 2017
726
727
SDL_stack_free(padding);
Sep 21, 2017
Sep 21, 2017
728
SDL_memcpy(cvt->buf, dst, cvt->len_cvt); /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
Jan 9, 2017
Jan 9, 2017
729
730
731
732
733
734
735
736
737
738
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. */
Nov 5, 2016
Nov 5, 2016
739
740
#define RESAMPLER_FUNCS(chans) \
static void SDLCALL \
Jan 9, 2017
Jan 9, 2017
741
742
SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
SDL_ResampleCVT(cvt, chans, format); \
Nov 5, 2016
Nov 5, 2016
743
744
745
746
747
748
749
750
}
RESAMPLER_FUNCS(1)
RESAMPLER_FUNCS(2)
RESAMPLER_FUNCS(4)
RESAMPLER_FUNCS(6)
RESAMPLER_FUNCS(8)
#undef RESAMPLER_FUNCS
Jan 6, 2017
Jan 6, 2017
751
static SDL_AudioFilter
Jan 9, 2017
Jan 9, 2017
752
ChooseCVTResampler(const int dst_channels)
Jan 6, 2017
Jan 6, 2017
753
{
Jan 9, 2017
Jan 9, 2017
754
755
756
757
758
759
760
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;
Jan 6, 2017
Jan 6, 2017
761
762
}
Jan 9, 2017
Jan 9, 2017
763
return NULL;
Jan 6, 2017
Jan 6, 2017
764
765
766
767
768
769
770
771
772
773
774
775
}
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. */
}
Jan 9, 2017
Jan 9, 2017
776
filter = ChooseCVTResampler(dst_channels);
Jan 6, 2017
Jan 6, 2017
777
778
779
if (filter == NULL) {
return SDL_SetError("No conversion available for these rates");
}
Sep 21, 2017
Sep 21, 2017
781
782
783
784
if (SDL_PrepareResampleFilter() < 0) {
return -1;
}
Jan 6, 2017
Jan 6, 2017
785
/* Update (cvt) with filter details... */
Jun 12, 2017
Jun 12, 2017
786
787
788
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
Sep 21, 2017
Sep 21, 2017
789
790
791
792
793
794
795
796
797
798
/* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator).
!!! FIXME in 2.1: We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
!!! FIXME in 2.1: so we steal the ninth and tenth slot. :( */
if (cvt->filter_index >= (SDL_AUDIOCVT_MAX_FILTERS-2)) {
return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS-2);
}
cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1] = (SDL_AudioFilter) (size_t) src_rate;
cvt->filters[SDL_AUDIOCVT_MAX_FILTERS] = (SDL_AudioFilter) (size_t) dst_rate;
Jan 6, 2017
Jan 6, 2017
799
800
801
802
803
804
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);
Sep 21, 2017
Sep 21, 2017
807
808
809
810
811
/* !!! FIXME: remove this if we can get the resampler to work in-place again. */
/* the buffer is big enough to hold the destination now, but
we need it large enough to hold a separate scratch buffer. */
cvt->len_mult *= 2;
Jan 6, 2017
Jan 6, 2017
812
return 1; /* added a converter. */
Jun 13, 2017
Jun 13, 2017
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
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 */
Aug 29, 2017
Aug 29, 2017
846
847
case 8: /* 7.1 */
return SDL_TRUE; /* supported. */
Jun 13, 2017
Jun 13, 2017
848
849
850
851
852
853
854
855
default:
break;
}
return SDL_FALSE; /* unsupported. */
}
856
857
/* Creates a set of audio filters to convert from one format to another.
Aug 18, 2017
Aug 18, 2017
858
859
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.
860
861
862
863
864
865
866
867
868
869
870
871
*/
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");
}
Jan 6, 2017
Jan 6, 2017
872
873
874
/* Make sure we zero out the audio conversion before error checking */
SDL_zerop(cvt);
Jun 13, 2017
Jun 13, 2017
875
if (!SDL_SupportedAudioFormat(src_fmt)) {
876
return SDL_SetError("Invalid source format");
Jun 13, 2017
Jun 13, 2017
877
} else if (!SDL_SupportedAudioFormat(dst_fmt)) {
878
return SDL_SetError("Invalid destination format");
Jun 13, 2017
Jun 13, 2017
879
880
881
882
883
884
885
886
} 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");
Nov 5, 2016
Nov 5, 2016
889
#if DEBUG_CONVERT
890
891
892
893
894
895
896
897
898
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;
Sep 21, 2017
Sep 21, 2017
899
SDL_zero(cvt->filters);
900
901
902
903
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
Aug 29, 2017
Aug 29, 2017
904
905
906
/* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
SDL_ChooseAudioConverters();
Nov 5, 2016
Nov 5, 2016
907
908
909
910
911
912
913
914
915
/* 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
Jan 6, 2017
Jan 6, 2017
916
buffer is likely to be CPU cache-friendly, avoiding the
Nov 5, 2016
Nov 5, 2016
917
918
919
920
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. */
Jan 6, 2017
Jan 6, 2017
921
922
923
924
925
926
927
928
/* 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)) {
Jun 12, 2017
Jun 12, 2017
929
930
931
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
Jan 6, 2017
Jan 6, 2017
932
933
934
cvt->needed = 1;
return 1;
}
Nov 5, 2016
Nov 5, 2016
935
936
}
937
/* Convert data types, if necessary. Updates (cvt). */
Jan 6, 2017
Jan 6, 2017
938
if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
939
940
941
942
return -1; /* shouldn't happen, but just in case... */
}
/* Channel conversion */
Aug 29, 2017
Aug 29, 2017
943
944
945
if (src_channels < dst_channels) {
/* Upmixing */
/* Mono -> Stereo [-> ...] */
946
if ((src_channels == 1) && (dst_channels > 1)) {
Jun 12, 2017
Jun 12, 2017
947
948
949
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
return -1;
}
950
951
952
953
cvt->len_mult *= 2;
src_channels = 2;
cvt->len_ratio *= 2;
}
Aug 29, 2017
Aug 29, 2017
954
955
/* [Mono ->] Stereo -> 5.1 [-> 7.1] */
if ((src_channels == 2) && (dst_channels >= 6)) {
Jun 12, 2017
Jun 12, 2017
956
957
958
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) {
return -1;
}
959
960
961
962
src_channels = 6;
cvt->len_mult *= 3;
cvt->len_ratio *= 3;
}
Aug 29, 2017
Aug 29, 2017
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
/* Quad -> 5.1 [-> 7.1] */
if ((src_channels == 4) && (dst_channels >= 6)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadTo51) < 0) {
return -1;
}
src_channels = 6;
cvt->len_mult = (cvt->len_mult * 3 + 1) / 2;
cvt->len_ratio *= 1.5;
}
/* [[Mono ->] Stereo ->] 5.1 -> 7.1 */
if ((src_channels == 6) && (dst_channels == 8)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51To71) < 0) {
return -1;
}
src_channels = 8;
cvt->len_mult = (cvt->len_mult * 4 + 2) / 3;
/* Should be numerically exact with every valid input to this
function */
cvt->len_ratio = cvt->len_ratio * 4 / 3;
}
/* [Mono ->] Stereo -> Quad */
984
if ((src_channels == 2) && (dst_channels == 4)) {
Jun 12, 2017
Jun 12, 2017
985
986
987
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoToQuad) < 0) {
return -1;
}
988
989
990
991
src_channels = 4;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
}
Aug 29, 2017
Aug 29, 2017
992
993
994
995
996
997
} else if (src_channels > dst_channels) {
/* Downmixing */
/* 7.1 -> 5.1 [-> Stereo [-> Mono]] */
/* 7.1 -> 5.1 [-> Quad] */
if ((src_channels == 8) && (dst_channels <= 6)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert71To51) < 0) {
Jun 12, 2017
Jun 12, 2017
998
999
return -1;
}
Aug 29, 2017
Aug 29, 2017
1000
src_channels = 6;