SDL_audiotypecvt.c
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/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 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"
#include "SDL_audio.h"
#include "SDL_audio_c.h"
#include "SDL_cpuinfo.h"
#include "SDL_assert.h"
/* !!! FIXME: write NEON code. */
#define HAVE_NEON_INTRINSICS 0
#ifdef __SSE2__
#define HAVE_SSE2_INTRINSICS 1
#endif
#if defined(__x86_64__) && HAVE_SSE2_INTRINSICS
#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* x86_64 guarantees SSE2. */
#elif __MACOSX__ && HAVE_SSE2_INTRINSICS
#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* Mac OS X/Intel guarantees SSE2. */
#elif defined(__ARM_ARCH) && (__ARM_ARCH >= 8) && HAVE_NEON_INTRINSICS
#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* ARMv8+ promise NEON. */
#elif defined(__APPLE__) && defined(__ARM_ARCH) && (__ARM_ARCH >= 7) && HAVE_NEON_INTRINSICS
#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* All Apple ARMv7 chips promise NEON support. */
#endif
/* Set to zero if platform is guaranteed to use a SIMD codepath here. */
#ifndef NEED_SCALAR_CONVERTER_FALLBACKS
#define NEED_SCALAR_CONVERTER_FALLBACKS 1
#endif
/* Function pointers set to a CPU-specific implementation. */
SDL_AudioFilter SDL_Convert_S8_to_F32 = NULL;
SDL_AudioFilter SDL_Convert_U8_to_F32 = NULL;
SDL_AudioFilter SDL_Convert_S16_to_F32 = NULL;
SDL_AudioFilter SDL_Convert_U16_to_F32 = NULL;
SDL_AudioFilter SDL_Convert_S32_to_F32 = NULL;
SDL_AudioFilter SDL_Convert_F32_to_S8 = NULL;
SDL_AudioFilter SDL_Convert_F32_to_U8 = NULL;
SDL_AudioFilter SDL_Convert_F32_to_S16 = NULL;
SDL_AudioFilter SDL_Convert_F32_to_U16 = NULL;
SDL_AudioFilter SDL_Convert_F32_to_S32 = NULL;
#define DIVBY128 0.0078125f
#define DIVBY32768 0.000030517578125f
#define DIVBY2147483648 0.00000000046566128730773926
#if NEED_SCALAR_CONVERTER_FALLBACKS
static void SDLCALL
SDL_Convert_S8_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint8 *src = ((const Sint8 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32");
for (i = cvt->len_cvt; i; --i, --src, --dst) {
*dst = ((float) *src) * DIVBY128;
}
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_U8_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Uint8 *src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32");
for (i = cvt->len_cvt; i; --i, --src, --dst) {
*dst = (((float) *src) * DIVBY128) - 1.0f;
}
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_S16_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint16 *src = ((const Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32");
for (i = cvt->len_cvt / sizeof (Sint16); i; --i, --src, --dst) {
*dst = ((float) *src) * DIVBY32768;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_U16_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Uint16 *src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32");
for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) {
*dst = (((float) *src) * DIVBY32768) - 1.0f;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_S32_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint32 *src = (const Sint32 *) cvt->buf;
float *dst = (float *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32");
for (i = cvt->len_cvt / sizeof (Sint32); i; --i, ++src, ++dst) {
*dst = (float) (((double) *src) * DIVBY2147483648);
}
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_S8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint8 *dst = (Sint8 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8");
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample > 1.0f) {
*dst = 127;
} else if (sample < -1.0f) {
*dst = -127;
} else {
*dst = (Sint8)(sample * 127.0f);
}
}
cvt->len_cvt /= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S8);
}
}
static void SDLCALL
SDL_Convert_F32_to_U8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Uint8 *dst = (Uint8 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8");
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample > 1.0f) {
*dst = 255;
} else if (sample < -1.0f) {
*dst = 0;
} else {
*dst = (Uint8)((sample + 1.0f) * 127.0f);
}
}
cvt->len_cvt /= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_U8);
}
}
static void SDLCALL
SDL_Convert_F32_to_S16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint16 *dst = (Sint16 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16");
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample > 1.0f) {
*dst = 32767;
} else if (sample < -1.0f) {
*dst = -32767;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_U16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Uint16 *dst = (Uint16 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16");
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample > 1.0f) {
*dst = 65534;
} else if (sample < -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
}
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_S32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint32 *dst = (Sint32 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32");
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample > 1.0f) {
*dst = 2147483647;
} else if (sample < -1.0f) {
*dst = -2147483647;
} else {
*dst = (Sint32)((double)sample * 2147483647.0);
}
}
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS);
}
}
#endif
#if HAVE_SSE2_INTRINSICS
static void SDLCALL
SDL_Convert_S8_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint8 *src = ((const Sint8 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32 (using SSE2)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt; i && (((size_t) (dst-15)) & 15); --i, --src, --dst) {
*dst = ((float) *src) * DIVBY128;
}
src -= 15; dst -= 15; /* adjust to read SSE blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128i *mmsrc = (const __m128i *) src;
const __m128i zero = _mm_setzero_si128();
const __m128 divby128 = _mm_set1_ps(DIVBY128);
while (i >= 16) { /* 16 * 8-bit */
const __m128i bytes = _mm_load_si128(mmsrc); /* get 16 sint8 into an XMM register. */
/* treat as int16, shift left to clear every other sint16, then back right with sign-extend. Now sint16. */
const __m128i shorts1 = _mm_srai_epi16(_mm_slli_epi16(bytes, 8), 8);
/* right-shift-sign-extend gets us sint16 with the other set of values. */
const __m128i shorts2 = _mm_srai_epi16(bytes, 8);
/* unpack against zero to make these int32, shift to make them sign-extend, convert to float, multiply. Whew! */
const __m128 floats1 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpacklo_epi16(shorts1, zero), 16), 16)), divby128);
const __m128 floats2 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpacklo_epi16(shorts2, zero), 16), 16)), divby128);
const __m128 floats3 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpackhi_epi16(shorts1, zero), 16), 16)), divby128);
const __m128 floats4 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpackhi_epi16(shorts2, zero), 16), 16)), divby128);
/* Interleave back into correct order, store. */
_mm_store_ps(dst, _mm_unpacklo_ps(floats1, floats2));
_mm_store_ps(dst+4, _mm_unpackhi_ps(floats1, floats2));
_mm_store_ps(dst+8, _mm_unpacklo_ps(floats3, floats4));
_mm_store_ps(dst+12, _mm_unpackhi_ps(floats3, floats4));
i -= 16; mmsrc--; dst -= 16;
}
src = (const Sint8 *) mmsrc;
}
src += 15; dst += 15; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = ((float) *src) * DIVBY128;
i--; src--; dst--;
}
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_U8_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Uint8 *src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32 (using SSE2)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt; i && (((size_t) (dst-15)) & 15); --i, --src, --dst) {
*dst = (((float) *src) * DIVBY128) - 1.0f;
}
src -= 15; dst -= 15; /* adjust to read SSE blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128i *mmsrc = (const __m128i *) src;
const __m128i zero = _mm_setzero_si128();
const __m128 divby128 = _mm_set1_ps(DIVBY128);
const __m128 minus1 = _mm_set1_ps(-1.0f);
while (i >= 16) { /* 16 * 8-bit */
const __m128i bytes = _mm_load_si128(mmsrc); /* get 16 uint8 into an XMM register. */
/* treat as int16, shift left to clear every other sint16, then back right with zero-extend. Now uint16. */
const __m128i shorts1 = _mm_srli_epi16(_mm_slli_epi16(bytes, 8), 8);
/* right-shift-zero-extend gets us uint16 with the other set of values. */
const __m128i shorts2 = _mm_srli_epi16(bytes, 8);
/* unpack against zero to make these int32, convert to float, multiply, add. Whew! */
/* Note that AVX2 can do floating point multiply+add in one instruction, fwiw. SSE2 cannot. */
const __m128 floats1 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi16(shorts1, zero)), divby128), minus1);
const __m128 floats2 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi16(shorts2, zero)), divby128), minus1);
const __m128 floats3 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi16(shorts1, zero)), divby128), minus1);
const __m128 floats4 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi16(shorts2, zero)), divby128), minus1);
/* Interleave back into correct order, store. */
_mm_store_ps(dst, _mm_unpacklo_ps(floats1, floats2));
_mm_store_ps(dst+4, _mm_unpackhi_ps(floats1, floats2));
_mm_store_ps(dst+8, _mm_unpacklo_ps(floats3, floats4));
_mm_store_ps(dst+12, _mm_unpackhi_ps(floats3, floats4));
i -= 16; mmsrc--; dst -= 16;
}
src = (const Uint8 *) mmsrc;
}
src += 15; dst += 15; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (((float) *src) * DIVBY128) - 1.0f;
i--; src--; dst--;
}
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_S16_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint16 *src = ((const Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32 (using SSE2)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt / sizeof (Sint16); i && (((size_t) (dst-7)) & 15); --i, --src, --dst) {
*dst = ((float) *src) * DIVBY32768;
}
src -= 7; dst -= 7; /* adjust to read SSE blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 divby32768 = _mm_set1_ps(DIVBY32768);
while (i >= 8) { /* 8 * 16-bit */
const __m128i ints = _mm_load_si128((__m128i const *) src); /* get 8 sint16 into an XMM register. */
/* treat as int32, shift left to clear every other sint16, then back right with sign-extend. Now sint32. */
const __m128i a = _mm_srai_epi32(_mm_slli_epi32(ints, 16), 16);
/* right-shift-sign-extend gets us sint32 with the other set of values. */
const __m128i b = _mm_srai_epi32(ints, 16);
/* Interleave these back into the right order, convert to float, multiply, store. */
_mm_store_ps(dst, _mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi32(a, b)), divby32768));
_mm_store_ps(dst+4, _mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi32(a, b)), divby32768));
i -= 8; src -= 8; dst -= 8;
}
}
src += 7; dst += 7; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = ((float) *src) * DIVBY32768;
i--; src--; dst--;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_U16_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Uint16 *src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32 (using SSE2)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt / sizeof (Sint16); i && (((size_t) (dst-7)) & 15); --i, --src, --dst) {
*dst = (((float) *src) * DIVBY32768) - 1.0f;
}
src -= 7; dst -= 7; /* adjust to read SSE blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 divby32768 = _mm_set1_ps(DIVBY32768);
const __m128 minus1 = _mm_set1_ps(1.0f);
while (i >= 8) { /* 8 * 16-bit */
const __m128i ints = _mm_load_si128((__m128i const *) src); /* get 8 sint16 into an XMM register. */
/* treat as int32, shift left to clear every other sint16, then back right with zero-extend. Now sint32. */
const __m128i a = _mm_srli_epi32(_mm_slli_epi32(ints, 16), 16);
/* right-shift-sign-extend gets us sint32 with the other set of values. */
const __m128i b = _mm_srli_epi32(ints, 16);
/* Interleave these back into the right order, convert to float, multiply, store. */
_mm_store_ps(dst, _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi32(a, b)), divby32768), minus1));
_mm_store_ps(dst+4, _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi32(a, b)), divby32768), minus1));
i -= 8; src -= 8; dst -= 8;
}
}
src += 7; dst += 7; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (((float) *src) * DIVBY32768) - 1.0f;
i--; src--; dst--;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
#if defined(__GNUC__) && (__GNUC__ < 4)
/* these were added as of gcc-4.0: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=19418 */
static inline __m128 _mm_castsi128_ps(__m128i __A) {
return (__m128) __A;
}
static inline __m128i _mm_castps_si128(__m128 __A) {
return (__m128i) __A;
}
#endif
static void SDLCALL
SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint32 *src = (const Sint32 *) cvt->buf;
float *dst = (float *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32 (using SSE2)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (Sint32); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = (float) (((double) *src) * DIVBY2147483648);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
SDL_assert(!i || ((((size_t) src) & 15) == 0));
{
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128d divby2147483648 = _mm_set1_pd(DIVBY2147483648);
const __m128i *mmsrc = (const __m128i *) src;
while (i >= 4) { /* 4 * sint32 */
const __m128i ints = _mm_load_si128(mmsrc);
/* bitshift the whole register over, so _mm_cvtepi32_pd can read the top ints in the bottom of the vector. */
const __m128d doubles1 = _mm_mul_pd(_mm_cvtepi32_pd(_mm_srli_si128(ints, 8)), divby2147483648);
const __m128d doubles2 = _mm_mul_pd(_mm_cvtepi32_pd(ints), divby2147483648);
/* convert to float32, bitshift/or to get these into a vector to store. */
_mm_store_ps(dst, _mm_castsi128_ps(_mm_or_si128(_mm_slli_si128(_mm_castps_si128(_mm_cvtpd_ps(doubles1)), 8), _mm_castps_si128(_mm_cvtpd_ps(doubles2)))));
i -= 4; mmsrc++; dst += 4;
}
src = (const Sint32 *) mmsrc;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (float) (((double) *src) * DIVBY2147483648);
i--; src++; dst++;
}
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_S8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint8 *dst = (Sint8 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8 (using SSE2)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = (Sint8) (*src * 127.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 mulby127 = _mm_set1_ps(127.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 16) { /* 16 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+4), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+8), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+12), mulby127)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_packs_epi16(_mm_packs_epi32(ints1, ints2), _mm_packs_epi32(ints3, ints4))); /* pack down, store out. */
i -= 16; src += 16; mmdst++;
}
dst = (Sint8 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Sint8) (*src * 127.0f);
i--; src++; dst++;
}
cvt->len_cvt /= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S8);
}
}
static void SDLCALL
SDL_Convert_F32_to_U8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Uint8 *dst = (Uint8 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8 (using SSE2)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = (Uint8) ((*src + 1.0f) * 127.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 add1 = _mm_set1_ps(1.0f);
const __m128 mulby127 = _mm_set1_ps(127.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 16) { /* 16 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src), add1), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src+4), add1), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src+8), add1), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src+12), add1), mulby127)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_packus_epi16(_mm_packs_epi32(ints1, ints2), _mm_packs_epi32(ints3, ints4))); /* pack down, store out. */
i -= 16; src += 16; mmdst++;
}
dst = (Uint8 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Uint8) ((*src + 1.0f) * 127.0f);
i--; src++; dst++;
}
cvt->len_cvt /= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_U8);
}
}
static void SDLCALL
SDL_Convert_F32_to_S16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint16 *dst = (Sint16 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16 (using SSE2)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = (Sint16) (*src * 32767.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 mulby32767 = _mm_set1_ps(32767.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 8) { /* 8 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src), mulby32767)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+4), mulby32767)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_packs_epi32(ints1, ints2)); /* pack to sint16, store out. */
i -= 8; src += 8; mmdst++;
}
dst = (Sint16 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Sint16) (*src * 32767.0f);
i--; src++; dst++;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_U16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Uint16 *dst = (Uint16 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16 (using SSE2)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = (Uint16) ((*src + 1.0f) * 32767.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
/* This calculates differently than the scalar path because SSE2 can't
pack int32 data down to unsigned int16. _mm_packs_epi32 does signed
saturation, so that would corrupt our data. _mm_packus_epi32 exists,
but not before SSE 4.1. So we convert from float to sint16, packing
that down with legit signed saturation, and then xor the top bit
against 1. This results in the correct unsigned 16-bit value, even
though it looks like dark magic. */
const __m128 mulby32767 = _mm_set1_ps(32767.0f);
const __m128i topbit = _mm_set1_epi16(-32768);
__m128i *mmdst = (__m128i *) dst;
while (i >= 8) { /* 8 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src), mulby32767)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+4), mulby32767)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_xor_si128(_mm_packs_epi32(ints1, ints2), topbit)); /* pack to sint16, xor top bit, store out. */
i -= 8; src += 8; mmdst++;
}
dst = (Uint16 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Uint16) ((*src + 1.0f) * 32767.0f);
i--; src++; dst++;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint32 *dst = (Sint32 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32 (using SSE2)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = (Sint32) (((double) *src) * 2147483647.0);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
SDL_assert(!i || ((((size_t) src) & 15) == 0));
{
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128d mulby2147483647 = _mm_set1_pd(2147483647.0);
__m128i *mmdst = (__m128i *) dst;
while (i >= 4) { /* 4 * float32 */
const __m128 floats = _mm_load_ps(src);
/* bitshift the whole register over, so _mm_cvtps_pd can read the top floats in the bottom of the vector. */
const __m128d doubles1 = _mm_mul_pd(_mm_cvtps_pd(_mm_castsi128_ps(_mm_srli_si128(_mm_castps_si128(floats), 8))), mulby2147483647);
const __m128d doubles2 = _mm_mul_pd(_mm_cvtps_pd(floats), mulby2147483647);
_mm_store_si128(mmdst, _mm_or_si128(_mm_slli_si128(_mm_cvtpd_epi32(doubles1), 8), _mm_cvtpd_epi32(doubles2)));
i -= 4; src += 4; mmdst++;
}
dst = (Sint32 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Sint32) (((double) *src) * 2147483647.0);
i--; src++; dst++;
}
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS);
}
}
#endif
void SDL_ChooseAudioConverters(void)
{
static SDL_bool converters_chosen = SDL_FALSE;
if (converters_chosen) {
return;
}
#define SET_CONVERTER_FUNCS(fntype) \
SDL_Convert_S8_to_F32 = SDL_Convert_S8_to_F32_##fntype; \
SDL_Convert_U8_to_F32 = SDL_Convert_U8_to_F32_##fntype; \
SDL_Convert_S16_to_F32 = SDL_Convert_S16_to_F32_##fntype; \
SDL_Convert_U16_to_F32 = SDL_Convert_U16_to_F32_##fntype; \
SDL_Convert_S32_to_F32 = SDL_Convert_S32_to_F32_##fntype; \
SDL_Convert_F32_to_S8 = SDL_Convert_F32_to_S8_##fntype; \
SDL_Convert_F32_to_U8 = SDL_Convert_F32_to_U8_##fntype; \
SDL_Convert_F32_to_S16 = SDL_Convert_F32_to_S16_##fntype; \
SDL_Convert_F32_to_U16 = SDL_Convert_F32_to_U16_##fntype; \
SDL_Convert_F32_to_S32 = SDL_Convert_F32_to_S32_##fntype; \
converters_chosen = SDL_TRUE
#if HAVE_SSE2_INTRINSICS
if (SDL_HasSSE2()) {
SET_CONVERTER_FUNCS(SSE2);
return;
}
#endif
#if NEED_SCALAR_CONVERTER_FALLBACKS
SET_CONVERTER_FUNCS(Scalar);
#endif
#undef SET_CONVERTER_FUNCS
SDL_assert(converters_chosen == SDL_TRUE);
}
/* vi: set ts=4 sw=4 expandtab: */
LOG_DEBUG_CONVERT
#define LOG_DEBUG_CONVERT(from, to)
Definition: SDL_audio_c.h:34
SDL_TRUE
Definition: SDL_stdinc.h:142
dst
GLenum GLenum dst
Definition: SDL_opengl_glext.h:1737
SDL_AudioCVT::buf
Uint8 * buf
Definition: SDL_audio.h:231
SDL_Convert_F32_to_S32_Scalar
static void SDL_Convert_F32_to_S32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:268
SDL_Convert_F32_to_U16
SDL_AudioFilter SDL_Convert_F32_to_U16
Definition: SDL_audiotypecvt.c:59
SDL_Convert_F32_to_S16
SDL_AudioFilter SDL_Convert_F32_to_S16
Definition: SDL_audiotypecvt.c:58
SDL_ChooseAudioConverters
void SDL_ChooseAudioConverters(void)
Definition: SDL_audiotypecvt.c:792
SDL_FALSE
Definition: SDL_stdinc.h:141
SDL_AudioCVT::filter_index
int filter_index
Definition: SDL_audio.h:237
SDL_Convert_U8_to_F32
SDL_AudioFilter SDL_Convert_U8_to_F32
Definition: SDL_audiotypecvt.c:52
SDL_Convert_S32_to_F32_Scalar
static void SDL_Convert_S32_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:146
f
GLfloat f
Definition: SDL_opengl_glext.h:1870
SDL_AudioFormat
Uint16 SDL_AudioFormat
Audio format flags.
Definition: SDL_audio.h:64
src
GLenum src
Definition: SDL_opengl_glext.h:1737
AUDIO_S16SYS
#define AUDIO_S16SYS
Definition: SDL_audio.h:123
SDL_Convert_F32_to_U8
SDL_AudioFilter SDL_Convert_F32_to_U8
Definition: SDL_audiotypecvt.c:57
SDL_AudioCVT
A structure to hold a set of audio conversion filters and buffers.
Definition: SDL_audio.h:225
SDL_Convert_U16_to_F32_Scalar
static void SDL_Convert_U16_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:127
SDL_cpuinfo.h
SDL_Convert_S16_to_F32
SDL_AudioFilter SDL_Convert_S16_to_F32
Definition: SDL_audiotypecvt.c:53
format
GLint GLint GLsizei GLsizei GLsizei GLint GLenum format
Definition: SDL_opengl.h:1572
size_t
unsigned int size_t
Definition: SDL_config_windows.h:68
zero
static const double zero
Definition: e_atan2.c:44
AUDIO_U8
#define AUDIO_U8
Definition: SDL_audio.h:89
SDL_AudioCVT::filters
SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS+1]
Definition: SDL_audio.h:236
AUDIO_F32SYS
#define AUDIO_F32SYS
Definition: SDL_audio.h:125
SDL_Convert_F32_to_U16_Scalar
static void SDL_Convert_F32_to_U16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:242
Sint8
int8_t Sint8
Definition: SDL_stdinc.h:151
SDL_Convert_S16_to_F32_Scalar
static void SDL_Convert_S16_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:108
SDL_audio_c.h
Uint8
uint8_t Uint8
Definition: SDL_stdinc.h:157
SDL_Convert_U8_to_F32_Scalar
static void SDL_Convert_U8_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:89
SDL_Convert_S8_to_F32_Scalar
static void SDL_Convert_S8_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:70
Sint32
int32_t Sint32
Definition: SDL_stdinc.h:175
SDL_Convert_F32_to_S8_Scalar
static void SDL_Convert_F32_to_S8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:164
SDL_AudioFilter
void(* SDL_AudioFilter)(struct SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audio.h:192
i
return Display return Display Bool Bool int int int return Display XEvent Bool(*) XPointer return Display return Display Drawable _Xconst char unsigned int unsigned int return Display Pixmap Pixmap XColor XColor unsigned int unsigned int return Display _Xconst char char int char return Display Visual unsigned int int int char unsigned int unsigned int in i)
Definition: SDL_x11sym.h:50
SDL_assert
#define SDL_assert(condition)
Definition: SDL_assert.h:169
SDL_AudioCVT::len_cvt
int len_cvt
Definition: SDL_audio.h:233
NULL
#define NULL
Definition: begin_code.h:164
SDL_bool
SDL_bool
Definition: SDL_stdinc.h:139
DIVBY2147483648
#define DIVBY2147483648
Definition: SDL_audiotypecvt.c:65
SDL_audio.h
SDL_Convert_F32_to_S16_Scalar
static void SDL_Convert_F32_to_S16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:216
SDL_HasSSE2
#define SDL_HasSSE2
Definition: SDL_dynapi_overrides.h:108
SDL_Convert_S8_to_F32
SDL_AudioFilter SDL_Convert_S8_to_F32
Definition: SDL_audiotypecvt.c:51
AUDIO_S32SYS
#define AUDIO_S32SYS
Definition: SDL_audio.h:124
Uint16
uint16_t Uint16
Definition: SDL_stdinc.h:169
SDL_Convert_F32_to_S32
SDL_AudioFilter SDL_Convert_F32_to_S32
Definition: SDL_audiotypecvt.c:60
SDL_Convert_F32_to_S8
SDL_AudioFilter SDL_Convert_F32_to_S8
Definition: SDL_audiotypecvt.c:56
SDL_Convert_F32_to_U8_Scalar
static void SDL_Convert_F32_to_U8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiotypecvt.c:190
DIVBY32768
#define DIVBY32768
Definition: SDL_audiotypecvt.c:64
DIVBY128
#define DIVBY128
Definition: SDL_audiotypecvt.c:63
SDL_Convert_U16_to_F32
SDL_AudioFilter SDL_Convert_U16_to_F32
Definition: SDL_audiotypecvt.c:54
a
GLboolean GLboolean GLboolean GLboolean a
Definition: SDL_opengl_glext.h:1109
AUDIO_S8
#define AUDIO_S8
Definition: SDL_audio.h:90
SDL_Convert_S32_to_F32
SDL_AudioFilter SDL_Convert_S32_to_F32
Definition: SDL_audiotypecvt.c:55
SDLCALL
#define SDLCALL
Definition: SDL_internal.h:45
b
GLboolean GLboolean GLboolean b
Definition: SDL_opengl_glext.h:1109
SET_CONVERTER_FUNCS
#define SET_CONVERTER_FUNCS(fntype)
AUDIO_U16SYS
#define AUDIO_U16SYS
Definition: SDL_audio.h:122
Sint16
int16_t Sint16
Definition: SDL_stdinc.h:163
SDL_assert.h