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1682 lines (1433 loc) · 57.3 KB
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/*
Simple DirectMedia Layer
Copyright (C) 1997-2021 Sam Lantinga <slouken@libsdl.org>
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 */
/* FIXME: Channel weights when converting from more channels to fewer may need to be adjusted, see https://msdn.microsoft.com/en-us/library/windows/desktop/ff819070(v=vs.85).aspx
*/
#include "SDL.h"
#include "SDL_audio.h"
#include "SDL_audio_c.h"
#include "SDL_loadso.h"
#include "../SDL_dataqueue.h"
#include "SDL_cpuinfo.h"
#define DEBUG_AUDIOSTREAM 0
#ifdef __SSE3__
#define HAVE_SSE3_INTRINSICS 1
#endif
#if HAVE_SSE3_INTRINSICS
/* Convert from stereo to mono. Average left and right. */
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
/* Convert from stereo to mono. Average left and right. */
static void SDLCALL
SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float *dst = (float *) cvt->buf;
const float *src = dst;
int i;
LOG_DEBUG_CONVERT("stereo", "mono");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
*(dst++) = (src[0] + src[1]) * 0.5f;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Convert from 5.1 to stereo. Average left and right, distribute center, discard LFE. */
static void SDLCALL
SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float *dst = (float *) cvt->buf;
const float *src = dst;
int i;
LOG_DEBUG_CONVERT("5.1", "stereo");
SDL_assert(format == AUDIO_F32SYS);
/* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
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 */
}
cvt->len_cvt /= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* 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 /= 2;
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. */
static void SDLCALL
SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
float *dst = (float *) cvt->buf;
const float *src = dst;
int i;
LOG_DEBUG_CONVERT("5.1", "quad");
SDL_assert(format == AUDIO_F32SYS);
/* SDL's 4.0 layout: FL+FR+BL+BR */
/* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
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 */
}
cvt->len_cvt /= 6;
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Upmix mono to stereo (by duplication) */
static void SDLCALL
SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
int i;
LOG_DEBUG_CONVERT("mono", "stereo");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / sizeof (float); i; --i) {
src--;
dst -= 2;
dst[0] = dst[1] = *src;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Upmix stereo to a pseudo-5.1 stream */
static void SDLCALL
SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
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);
for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
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] = lf; /* BL */
dst[5] = rf; /* BR */
}
cvt->len_cvt *= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* 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) */
static void SDLCALL
SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
float lf, rf;
int i;
LOG_DEBUG_CONVERT("stereo", "quad");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
dst[0] = lf; /* FL */
dst[1] = rf; /* FR */
dst[2] = lf; /* BL */
dst[3] = rf; /* BR */
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* 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);
}
}
/* SDL's resampler uses a "bandlimited interpolation" algorithm:
https://ccrma.stanford.edu/~jos/resample/ */
#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)
/* 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;
}
i0 += diff;
i++;
f *= (double) i;
}
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;
}
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;
}
static SDL_SpinLock ResampleFilterSpinlock = 0;
static float *ResamplerFilter = NULL;
static float *ResamplerFilterDifference = NULL;
int
SDL_PrepareResampleFilter(void)
{
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();
}
ResamplerFilterDifference = (float *) SDL_malloc(alloclen);
if (!ResamplerFilterDifference) {
SDL_free(ResamplerFilter);
ResamplerFilter = NULL;
SDL_AtomicUnlock(&ResampleFilterSpinlock);
return SDL_OutOfMemory();
}
kaiser_and_sinc(ResamplerFilter, ResamplerFilterDifference, RESAMPLER_FILTER_SIZE, beta);
}
SDL_AtomicUnlock(&ResampleFilterSpinlock);
return 0;
}
void
SDL_FreeResampleFilter(void)
{
SDL_free(ResamplerFilter);
SDL_free(ResamplerFilterDifference);
ResamplerFilter = NULL;
ResamplerFilterDifference = NULL;
}
static int
ResamplerPadding(const int inrate, const int outrate)
{
if (inrate == outrate) {
return 0;
} else if (inrate > outrate) {
return (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
}
return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
}
/* lpadding and rpadding are expected to be buffers of (ResamplePadding(inrate, outrate) * chans * sizeof (float)) bytes. */
static int
SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
const float *lpadding, const float *rpadding,
const float *inbuf, const int inbuflen,
float *outbuf, const int outbuflen)
{
const double finrate = (double) inrate;
const double outtimeincr = 1.0 / ((float) outrate);
const double ratio = ((float) outrate) / ((float) inrate);
const int paddinglen = ResamplerPadding(inrate, outrate);
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 = SDL_min(wantedoutframes, maxoutframes);
float *dst = outbuf;
double outtime = 0.0;
int i, j, chan;
for (i = 0; i < outframes; i++) {
const int srcindex = (int) (outtime * inrate);
const double intime = ((double) srcindex) / finrate;
const double innexttime = ((double) (srcindex + 1)) / finrate;
const double interpolation1 = 1.0 - ((innexttime - outtime) / (innexttime - intime));
const int filterindex1 = (int) (interpolation1 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
const double interpolation2 = 1.0 - interpolation1;
const int filterindex2 = (int) (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;
/* !!! 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];
outsample += (float)(insample * (ResamplerFilter[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation1 * ResamplerFilterDifference[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
}
for (j = 0; (filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
const int srcframe = srcindex + 1 + j;
/* !!! 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];
outsample += (float)(insample * (ResamplerFilter[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation2 * ResamplerFilterDifference[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
}
*(dst++) = outsample;
}
outtime += outtimeincr;
}
return outframes * chans * sizeof (float);
}
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) {
return SDL_SetError("No buffer allocated for conversion");
}
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if (cvt->filters[0] == NULL) {
return 0;
}
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0] (cvt, cvt->src_format);
return 0;
}
static void SDLCALL
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
#if DEBUG_CONVERT
printf("Converting byte order\n");
#endif
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;
}
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);
}
}
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;
}
static int
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
{
int retval = 0; /* 0 == no conversion necessary. */
if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
retval = 1; /* added a converter. */
}
if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
const Uint16 dst_bitsize = 32;
SDL_AudioFilter filter = NULL;
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;
case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
if (!filter) {
return SDL_SetError("No conversion from source format to float available");
}
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
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. */
}
return retval;
}
static int
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
{
int retval = 0; /* 0 == no conversion necessary. */
if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
const Uint16 src_bitsize = 32;
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;
case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
if (!filter) {
return SDL_SetError("No conversion from float to format 0x%.4x available", dst_fmt);
}
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
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)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
retval = 1; /* added a converter. */
}
return retval;
}
static void
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
{
/* !!! 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. :( */
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];
const float *src = (const float *) cvt->buf;
const int srclen = cvt->len_cvt;
/*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;
const int requestedpadding = ResamplerPadding(inrate, outrate);
int paddingsamples;
float *padding;
if (requestedpadding < SDL_MAX_SINT32 / chans) {
paddingsamples = requestedpadding * chans;
} else {
paddingsamples = 0;
}
SDL_assert(format == AUDIO_F32SYS);
/* we keep no streaming state here, so pad with silence on both ends. */
padding = (float *) SDL_calloc(paddingsamples ? paddingsamples : 1, sizeof (float));
if (!padding) {
SDL_OutOfMemory();
return;
}
cvt->len_cvt = SDL_ResampleAudio(chans, inrate, outrate, padding, padding, src, srclen, dst, dstlen);
SDL_free(padding);
SDL_memmove(cvt->buf, dst, cvt->len_cvt); /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
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. */
#define RESAMPLER_FUNCS(chans) \
static void SDLCALL \
SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
SDL_ResampleCVT(cvt, chans, format); \
}
RESAMPLER_FUNCS(1)
RESAMPLER_FUNCS(2)
RESAMPLER_FUNCS(4)
RESAMPLER_FUNCS(6)
RESAMPLER_FUNCS(8)
#undef RESAMPLER_FUNCS
static SDL_AudioFilter
ChooseCVTResampler(const int dst_channels)
{
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;
}
return NULL;
}
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. */
}
filter = ChooseCVTResampler(dst_channels);
if (filter == NULL) {
return SDL_SetError("No conversion available for these rates");
}
if (SDL_PrepareResampleFilter() < 0) {
return -1;
}
/* Update (cvt) with filter details... */
if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
return -1;
}
/* !!! 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;
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);
}
/* !!! 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;
return 1; /* added a converter. */
}
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 */
case 8: /* 7.1 */
return SDL_TRUE; /* supported. */
default:
break;
}
return SDL_FALSE; /* unsupported. */
}
/* Creates a set of audio filters to convert from one format to another.
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.
*/
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");
}
/* Make sure we zero out the audio conversion before error checking */
SDL_zerop(cvt);
if (!SDL_SupportedAudioFormat(src_fmt)) {
return SDL_SetError("Invalid source format");
} else if (!SDL_SupportedAudioFormat(dst_fmt)) {
return SDL_SetError("Invalid destination format");
} 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 equal to or less than zero");
} else if (dst_rate <= 0) {
return SDL_SetError("Destination rate is equal to or less than zero");
} else if (src_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
return SDL_SetError("Source rate is too high");
} else if (dst_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
return SDL_SetError("Destination rate is too high");
}
#if DEBUG_CONVERT
printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
#endif
/* Start off with no conversion necessary */
cvt->src_format = src_fmt;
cvt->dst_format = dst_fmt;
cvt->needed = 0;
cvt->filter_index = 0;
SDL_zeroa(cvt->filters);
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
/* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
SDL_ChooseAudioConverters();
/* 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
buffer is likely to be CPU cache-friendly, avoiding the
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. */
/* 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)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
return -1;
}
cvt->needed = 1;
return 1;
}
}
/* Convert data types, if necessary. Updates (cvt). */
if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
return -1; /* shouldn't happen, but just in case... */
}
/* Channel conversion */
if (src_channels < dst_channels) {
/* Upmixing */
/* Mono -> Stereo [-> ...] */
if ((src_channels == 1) && (dst_channels > 1)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
return -1;
}
cvt->len_mult *= 2;
src_channels = 2;
cvt->len_ratio *= 2;
}
/* [Mono ->] Stereo -> 5.1 [-> 7.1] */
if ((src_channels == 2) && (dst_channels >= 6)) {
if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) {
return -1;
}
src_channels = 6;
cvt->len_mult *= 3;
cvt->len_ratio *= 3;
}
/* 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;
}