alphatau 0.1.0

data/ext/alphatau/native/sdl/src/audio/SDL_audiodev.c

@@ -0,0 +1,124 @@
+/*
+  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"
+
+/* Get the name of the audio device we use for output */
+
+#if SDL_AUDIO_DRIVER_NETBSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO
+
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h> /* For close() */
+
+#include "SDL_stdinc.h"
+#include "SDL_audiodev_c.h"
+
+#ifndef _PATH_DEV_DSP
+#if defined(__NETBSD__) || defined(__OPENBSD__)
+#define _PATH_DEV_DSP  "/dev/audio"
+#else
+#define _PATH_DEV_DSP  "/dev/dsp"
+#endif
+#endif
+#ifndef _PATH_DEV_DSP24
+#define _PATH_DEV_DSP24 "/dev/sound/dsp"
+#endif
+#ifndef _PATH_DEV_AUDIO
+#define _PATH_DEV_AUDIO "/dev/audio"
+#endif
+
+static void
+test_device(const int iscapture, const char *fname, int flags, int (*test) (int fd))
+{
+    struct stat sb;
+    if ((stat(fname, &sb) == 0) && (S_ISCHR(sb.st_mode))) {
+        const int audio_fd = open(fname, flags, 0);
+        if (audio_fd >= 0) {
+            const int okay = test(audio_fd);
+            close(audio_fd);
+            if (okay) {
+                static size_t dummyhandle = 0;
+                dummyhandle++;
+                SDL_assert(dummyhandle != 0);
+                SDL_AddAudioDevice(iscapture, fname, (void *) dummyhandle);
+            }
+        }
+    }
+}
+
+static int
+test_stub(int fd)
+{
+    return 1;
+}
+
+static void
+SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*test)(int))
+{
+    const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
+    const char *audiodev;
+    char audiopath[1024];
+
+    if (test == NULL)
+        test = test_stub;
+
+    /* Figure out what our audio device is */
+    if (((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) &&
+        ((audiodev = SDL_getenv("AUDIODEV")) == NULL)) {
+        if (classic) {
+            audiodev = _PATH_DEV_AUDIO;
+        } else {
+            struct stat sb;
+
+            /* Added support for /dev/sound/\* in Linux 2.4 */
+            if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode))
+                && ((stat(_PATH_DEV_DSP24, &sb) == 0)
+                    && S_ISCHR(sb.st_mode))) {
+                audiodev = _PATH_DEV_DSP24;
+            } else {
+                audiodev = _PATH_DEV_DSP;
+            }
+        }
+    }
+    test_device(iscapture, audiodev, flags, test);
+
+    if (SDL_strlen(audiodev) < (sizeof(audiopath) - 3)) {
+        int instance = 0;
+        while (instance <= 64) {
+            SDL_snprintf(audiopath, SDL_arraysize(audiopath),
+                         "%s%d", audiodev, instance);
+            instance++;
+            test_device(iscapture, audiopath, flags, test);
+        }
+    }
+}
+
+void
+SDL_EnumUnixAudioDevices(const int classic, int (*test)(int))
+{
+    SDL_EnumUnixAudioDevices_Internal(SDL_TRUE, classic, test);
+    SDL_EnumUnixAudioDevices_Internal(SDL_FALSE, classic, test);
+}
+
+#endif /* Audio driver selection */
+
+/* vi: set ts=4 sw=4 expandtab: */

data/ext/alphatau/native/sdl/src/audio/SDL_audiodev_c.h

@@ -0,0 +1,38 @@
+/*
+  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.h"
+#include "../SDL_internal.h"
+#include "SDL_sysaudio.h"
+
+/* Open the audio device for playback, and don't block if busy */
+/* #define USE_BLOCKING_WRITES */
+
+#ifdef USE_BLOCKING_WRITES
+#define OPEN_FLAGS_OUTPUT O_WRONLY
+#define OPEN_FLAGS_INPUT O_RDONLY
+#else
+#define OPEN_FLAGS_OUTPUT (O_WRONLY|O_NONBLOCK)
+#define OPEN_FLAGS_INPUT (O_RDONLY|O_NONBLOCK)
+#endif
+
+extern void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int));
+
+/* vi: set ts=4 sw=4 expandtab: */

data/ext/alphatau/native/sdl/src/audio/SDL_audiotypecvt.c

@@ -0,0 +1,1430 @@
+/*
+  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"
+
+#ifdef __ARM_NEON__
+#define HAVE_NEON_INTRINSICS 1
+#endif
+
+#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 DIVBY8388607 0.00000011920930376163766f
+
+
+#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) (*src>>8)) * DIVBY8388607;
+    }
+
+    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 = -128;
+        } 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 = -32768;
+        } 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 = 65535;
+        } 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 = (Sint32) -2147483648LL;
+        } else {
+            *dst = ((Sint32)(sample * 8388607.0f)) << 8;
+        }
+    }
+
+    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);
+    }
+}
+
+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) (*src>>8)) * DIVBY8388607;
+    }
+
+    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 __m128 divby8388607 = _mm_set1_ps(DIVBY8388607);
+        const __m128i *mmsrc = (const __m128i *) src;
+        while (i >= 4) {   /* 4 * sint32 */
+            /* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */
+            _mm_store_ps(dst, _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_load_si128(mmsrc), 8)), divby8388607));
+            i -= 4; mmsrc++; dst += 4;
+        }
+        src = (const Sint32 *) mmsrc;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        *dst = ((float) (*src>>8)) * DIVBY8388607;
+        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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 127;
+        } else if (sample <= -1.0f) {
+            *dst = -128;
+        } else {
+            *dst = (Sint8)(sample * 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 one = _mm_set1_ps(1.0f);
+        const __m128 negone = _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_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+8)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+12)), one), mulby127));  /* load 4 floats, clamp, 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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 127;
+        } else if (sample <= -1.0f) {
+            *dst = -128;
+        } else {
+            *dst = (Sint8)(sample * 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) {
+        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);
+        }
+    }
+
+    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 one = _mm_set1_ps(1.0f);
+        const __m128 negone = _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_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+8)), one), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+12)), one), one), mulby127));  /* load 4 floats, clamp, 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) {
+        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);
+        }
+        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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 32767;
+        } else if (sample <= -1.0f) {
+            *dst = -32768;
+        } else {
+            *dst = (Sint16)(sample * 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 one = _mm_set1_ps(1.0f);
+        const __m128 negone = _mm_set1_ps(-1.0f);
+        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_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby32767));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), mulby32767));  /* load 4 floats, clamp, 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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 32767;
+        } else if (sample <= -1.0f) {
+            *dst = -32768;
+        } else {
+            *dst = (Sint16)(sample * 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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 65535;
+        } else if (sample <= -1.0f) {
+            *dst = 0;
+        } else {
+            *dst = (Uint16)((sample + 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);
+        const __m128 one = _mm_set1_ps(1.0f);
+        const __m128 negone = _mm_set1_ps(-1.0f);
+        __m128i *mmdst = (__m128i *) dst;
+        while (i >= 8) {   /* 8 * float32 */
+            const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby32767));  /* load 4 floats, clamp, convert to sint32 */
+            const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), mulby32767));  /* load 4 floats, clamp, 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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 65535;
+        } else if (sample <= -1.0f) {
+            *dst = 0;
+        } else {
+            *dst = (Uint16)((sample + 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) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 2147483647;
+        } else if (sample <= -1.0f) {
+            *dst = (Sint32) -2147483648LL;
+        } else {
+            *dst = ((Sint32)(sample * 8388607.0f)) << 8;
+        }
+    }
+
+    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 __m128 one = _mm_set1_ps(1.0f);
+        const __m128 negone = _mm_set1_ps(-1.0f);
+        const __m128 mulby8388607 = _mm_set1_ps(8388607.0f);
+        __m128i *mmdst = (__m128i *) dst;
+        while (i >= 4) {   /* 4 * float32 */
+            _mm_store_si128(mmdst, _mm_slli_epi32(_mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby8388607)), 8));  /* load 4 floats, clamp, convert to sint32 */
+            i -= 4; src += 4; mmdst++;
+        }
+        dst = (Sint32 *) mmdst;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 2147483647;
+        } else if (sample <= -1.0f) {
+            *dst = (Sint32) -2147483648LL;
+        } else {
+            *dst = ((Sint32)(sample * 8388607.0f)) << 8;
+        }
+        i--; src++; dst++;
+    }
+
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS);
+    }
+}
+#endif
+
+
+#if HAVE_NEON_INTRINSICS
+static void SDLCALL
+SDL_Convert_S8_to_F32_NEON(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 NEON)");
+
+    /* 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 NEON 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 NEON blocks as long as we have 16 bytes available. */
+        const int8_t *mmsrc = (const int8_t *) src;
+        const float32x4_t divby128 = vdupq_n_f32(DIVBY128);
+        while (i >= 16) {   /* 16 * 8-bit */
+            const int8x16_t bytes = vld1q_s8(mmsrc);  /* get 16 sint8 into a NEON register. */
+            const int16x8_t int16hi = vmovl_s8(vget_high_s8(bytes));  /* convert top 8 bytes to 8 int16 */
+            const int16x8_t int16lo = vmovl_s8(vget_low_s8(bytes));   /* convert bottom 8 bytes to 8 int16 */
+            /* split int16 to two int32, then convert to float, then multiply to normalize, store. */
+            vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16hi))), divby128));
+            vst1q_f32(dst+4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16hi))), divby128));
+            vst1q_f32(dst+8, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16lo))), divby128));
+            vst1q_f32(dst+12, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16lo))), divby128));
+            i -= 16; mmsrc -= 16; 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_NEON(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 NEON)");
+
+    /* 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 NEON 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 NEON blocks as long as we have 16 bytes available. */
+        const uint8_t *mmsrc = (const uint8_t *) src;
+        const float32x4_t divby128 = vdupq_n_f32(DIVBY128);
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        while (i >= 16) {   /* 16 * 8-bit */
+            const uint8x16_t bytes = vld1q_u8(mmsrc);  /* get 16 uint8 into a NEON register. */
+            const uint16x8_t uint16hi = vmovl_u8(vget_high_u8(bytes));  /* convert top 8 bytes to 8 uint16 */
+            const uint16x8_t uint16lo = vmovl_u8(vget_low_u8(bytes));   /* convert bottom 8 bytes to 8 uint16 */
+            /* split uint16 to two uint32, then convert to float, then multiply to normalize, subtract to adjust for sign, store. */
+            vst1q_f32(dst, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16hi))), divby128, one));
+            vst1q_f32(dst+4, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16hi))), divby128, one));
+            vst1q_f32(dst+8, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16lo))), divby128, one));
+            vst1q_f32(dst+12, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16lo))), divby128, one));
+            i -= 16; mmsrc -= 16; 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_NEON(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 NEON)");
+
+    /* 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 NEON 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 NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768);
+        while (i >= 8) {   /* 8 * 16-bit */
+            const int16x8_t ints = vld1q_s16((int16_t const *) src);  /* get 8 sint16 into a NEON register. */
+            /* split int16 to two int32, then convert to float, then multiply to normalize, store. */
+            vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(ints))), divby32768));
+            vst1q_f32(dst+4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(ints))), 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_NEON(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 NEON)");
+
+    /* 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 NEON 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 NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768);
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        while (i >= 8) {   /* 8 * 16-bit */
+            const uint16x8_t uints = vld1q_u16((uint16_t const *) src);  /* get 8 uint16 into a NEON register. */
+            /* split uint16 to two int32, then convert to float, then multiply to normalize, subtract for sign, store. */
+            vst1q_f32(dst, vmlsq_f32(one, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uints))), divby32768));
+            vst1q_f32(dst+4, vmlsq_f32(one, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uints))), 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) - 1.0f;
+        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_S32_to_F32_NEON(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 NEON)");
+
+    /* Get dst aligned to 16 bytes */
+    for (i = cvt->len_cvt / sizeof (Sint32); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
+        *dst = ((float) (*src>>8)) * DIVBY8388607;
+    }
+
+    SDL_assert(!i || ((((size_t) dst) & 15) == 0));
+    SDL_assert(!i || ((((size_t) src) & 15) == 0));
+
+    {
+        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t divby8388607 = vdupq_n_f32(DIVBY8388607);
+        const int32_t *mmsrc = (const int32_t *) src;
+        while (i >= 4) {   /* 4 * sint32 */
+            /* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */
+            vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vshrq_n_s32(vld1q_s32(mmsrc), 8)), divby8388607));
+            i -= 4; mmsrc += 4; dst += 4;
+        }
+        src = (const Sint32 *) mmsrc;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        *dst = ((float) (*src>>8)) * DIVBY8388607;
+        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_NEON(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 NEON)");
+
+    /* Get dst aligned to 16 bytes */
+    for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 127;
+        } else if (sample <= -1.0f) {
+            *dst = -128;
+        } else {
+            *dst = (Sint8)(sample * 127.0f);
+        }
+    }
+
+    SDL_assert(!i || ((((size_t) dst) & 15) == 0));
+
+    /* Make sure src is aligned too. */
+    if ((((size_t) src) & 15) == 0) {
+        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        const float32x4_t negone = vdupq_n_f32(-1.0f);
+        const float32x4_t mulby127 = vdupq_n_f32(127.0f);
+        int8_t *mmdst = (int8_t *) dst;
+        while (i >= 16) {   /* 16 * float32 */
+            const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const int32x4_t ints3 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+8)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const int32x4_t ints4 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+12)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
+            const int8x8_t i8lo = vmovn_s16(vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2))); /* narrow to sint16, combine, narrow to sint8 */
+            const int8x8_t i8hi = vmovn_s16(vcombine_s16(vmovn_s32(ints3), vmovn_s32(ints4))); /* narrow to sint16, combine, narrow to sint8 */
+            vst1q_s8(mmdst, vcombine_s8(i8lo, i8hi));  /* combine to int8x16_t, store out */
+            i -= 16; src += 16; mmdst += 16;
+        }
+        dst = (Sint8 *) mmdst;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 127;
+        } else if (sample <= -1.0f) {
+            *dst = -128;
+        } else {
+            *dst = (Sint8)(sample * 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_NEON(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 NEON)");
+
+    /* Get dst aligned to 16 bytes */
+    for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --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);
+        }
+    }
+
+    SDL_assert(!i || ((((size_t) dst) & 15) == 0));
+
+    /* Make sure src is aligned too. */
+    if ((((size_t) src) & 15) == 0) {
+        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        const float32x4_t negone = vdupq_n_f32(-1.0f);
+        const float32x4_t mulby127 = vdupq_n_f32(127.0f);
+        uint8_t *mmdst = (uint8_t *) dst;
+        while (i >= 16) {   /* 16 * float32 */
+            const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby127));  /* load 4 floats, clamp, convert to uint32 */
+            const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), one), mulby127));  /* load 4 floats, clamp, convert to uint32 */
+            const uint32x4_t uints3 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+8)), one), one), mulby127));  /* load 4 floats, clamp, convert to uint32 */
+            const uint32x4_t uints4 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+12)), one), one), mulby127));  /* load 4 floats, clamp, convert to uint32 */
+            const uint8x8_t ui8lo = vmovn_u16(vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2))); /* narrow to uint16, combine, narrow to uint8 */
+            const uint8x8_t ui8hi = vmovn_u16(vcombine_u16(vmovn_u32(uints3), vmovn_u32(uints4))); /* narrow to uint16, combine, narrow to uint8 */
+            vst1q_u8(mmdst, vcombine_u8(ui8lo, ui8hi));  /* combine to uint8x16_t, store out */
+            i -= 16; src += 16; mmdst += 16;
+        }
+
+        dst = (Uint8 *) mmdst;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        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);
+        }
+        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_NEON(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 NEON)");
+
+    /* Get dst aligned to 16 bytes */
+    for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 32767;
+        } else if (sample <= -1.0f) {
+            *dst = -32768;
+        } else {
+            *dst = (Sint16)(sample * 32767.0f);
+        }
+    }
+
+    SDL_assert(!i || ((((size_t) dst) & 15) == 0));
+
+    /* Make sure src is aligned too. */
+    if ((((size_t) src) & 15) == 0) {
+        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        const float32x4_t negone = vdupq_n_f32(-1.0f);
+        const float32x4_t mulby32767 = vdupq_n_f32(32767.0f);
+        int16_t *mmdst = (int16_t *) dst;
+        while (i >= 8) {   /* 8 * float32 */
+            const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby32767));  /* load 4 floats, clamp, convert to sint32 */
+            const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), mulby32767));  /* load 4 floats, clamp, convert to sint32 */
+            vst1q_s16(mmdst, vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2)));  /* narrow to sint16, combine, store out. */
+            i -= 8; src += 8; mmdst += 8;
+        }
+        dst = (Sint16 *) mmdst;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 32767;
+        } else if (sample <= -1.0f) {
+            *dst = -32768;
+        } else {
+            *dst = (Sint16)(sample * 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_NEON(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 NEON)");
+
+    /* Get dst aligned to 16 bytes */
+    for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 65535;
+        } else if (sample <= -1.0f) {
+            *dst = 0;
+        } else {
+            *dst = (Uint16)((sample + 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 NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        const float32x4_t negone = vdupq_n_f32(-1.0f);
+        const float32x4_t mulby32767 = vdupq_n_f32(32767.0f);
+        uint16_t *mmdst = (uint16_t *) dst;
+        while (i >= 8) {   /* 8 * float32 */
+            const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby32767));  /* load 4 floats, clamp, convert to uint32 */
+            const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), one), mulby32767));  /* load 4 floats, clamp, convert to uint32 */
+            vst1q_u16(mmdst, vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2)));  /* narrow to uint16, combine, store out. */
+            i -= 8; src += 8; mmdst += 8;
+        }
+        dst = (Uint16 *) mmdst;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 65535;
+        } else if (sample <= -1.0f) {
+            *dst = 0;
+        } else {
+            *dst = (Uint16)((sample + 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_NEON(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 NEON)");
+
+    /* Get dst aligned to 16 bytes */
+    for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 2147483647;
+        } else if (sample <= -1.0f) {
+            *dst = -2147483648;
+        } else {
+            *dst = ((Sint32)(sample * 8388607.0f)) << 8;
+        }
+    }
+
+    SDL_assert(!i || ((((size_t) dst) & 15) == 0));
+    SDL_assert(!i || ((((size_t) src) & 15) == 0));
+
+    {
+        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        const float32x4_t one = vdupq_n_f32(1.0f);
+        const float32x4_t negone = vdupq_n_f32(-1.0f);
+        const float32x4_t mulby8388607 = vdupq_n_f32(8388607.0f);
+        int32_t *mmdst = (int32_t *) dst;
+        while (i >= 4) {   /* 4 * float32 */
+            vst1q_s32(mmdst, vshlq_n_s32(vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby8388607)), 8));
+            i -= 4; src += 4; mmdst += 4;
+        }
+        dst = (Sint32 *) mmdst;
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        const float sample = *src;
+        if (sample >= 1.0f) {
+            *dst = 2147483647;
+        } else if (sample <= -1.0f) {
+            *dst = -2147483648;
+        } else {
+            *dst = ((Sint32)(sample * 8388607.0f)) << 8;
+        }
+        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 HAVE_NEON_INTRINSICS
+    if (SDL_HasNEON()) {
+        SET_CONVERTER_FUNCS(NEON);
+        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: */