alphatau 0.1.0

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

@@ -0,0 +1,79 @@
+/*
+  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.
+*/
+
+#ifndef SDL_audio_c_h_
+#define SDL_audio_c_h_
+
+#include "../SDL_internal.h"
+
+#ifndef DEBUG_CONVERT
+#define DEBUG_CONVERT 0
+#endif
+
+#if DEBUG_CONVERT
+#define LOG_DEBUG_CONVERT(from, to) fprintf(stderr, "Converting %s to %s.\n", from, to);
+#else
+#define LOG_DEBUG_CONVERT(from, to)
+#endif
+
+/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
+
+#ifdef HAVE_LIBSAMPLERATE_H
+#include "samplerate.h"
+extern SDL_bool SRC_available;
+extern int SRC_converter;
+extern SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error);
+extern int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data);
+extern int (*SRC_src_reset)(SRC_STATE *state);
+extern SRC_STATE* (*SRC_src_delete)(SRC_STATE *state);
+extern const char* (*SRC_src_strerror)(int error);
+#endif
+
+/* Functions to get a list of "close" audio formats */
+extern SDL_AudioFormat SDL_FirstAudioFormat(SDL_AudioFormat format);
+extern SDL_AudioFormat SDL_NextAudioFormat(void);
+
+/* Function to calculate the size and silence for a SDL_AudioSpec */
+extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
+
+/* Choose the audio filter functions below */
+extern void SDL_ChooseAudioConverters(void);
+
+/* These pointers get set during SDL_ChooseAudioConverters() to various SIMD implementations. */
+extern SDL_AudioFilter SDL_Convert_S8_to_F32;
+extern SDL_AudioFilter SDL_Convert_U8_to_F32;
+extern SDL_AudioFilter SDL_Convert_S16_to_F32;
+extern SDL_AudioFilter SDL_Convert_U16_to_F32;
+extern SDL_AudioFilter SDL_Convert_S32_to_F32;
+extern SDL_AudioFilter SDL_Convert_F32_to_S8;
+extern SDL_AudioFilter SDL_Convert_F32_to_U8;
+extern SDL_AudioFilter SDL_Convert_F32_to_S16;
+extern SDL_AudioFilter SDL_Convert_F32_to_U16;
+extern SDL_AudioFilter SDL_Convert_F32_to_S32;
+
+/* You need to call SDL_PrepareResampleFilter() before using the internal resampler.
+   SDL_AudioQuit() calls SDL_FreeResamplerFilter(), you should never call it yourself. */
+extern int SDL_PrepareResampleFilter(void);
+extern void SDL_FreeResampleFilter(void);
+
+#endif /* SDL_audio_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

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

@@ -0,0 +1,1673 @@
+/*
+  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"
+
+/* Functions for audio drivers to perform runtime conversion of audio format */
+
+/* FIXME: Channel weights when converting from more channels to fewer may need to be adjusted, see https://msdn.microsoft.com/en-us/library/windows/desktop/ff819070(v=vs.85).aspx
+*/
+
+#include "SDL.h"
+#include "SDL_audio.h"
+#include "SDL_audio_c.h"
+
+#include "SDL_loadso.h"
+#include "SDL_assert.h"
+#include "../SDL_dataqueue.h"
+#include "SDL_cpuinfo.h"
+
+#define DEBUG_AUDIOSTREAM 0
+
+#ifdef __SSE3__
+#define HAVE_SSE3_INTRINSICS 1
+#endif
+
+#if HAVE_SSE3_INTRINSICS
+/* Convert from stereo to mono. Average left and right. */
+static void SDLCALL
+SDL_ConvertStereoToMono_SSE3(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float *dst = (float *) cvt->buf;
+    const float *src = dst;
+    int i = cvt->len_cvt / 8;
+
+    LOG_DEBUG_CONVERT("stereo", "mono (using SSE3)");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    /* We can only do this if dst is aligned to 16 bytes; since src is the
+       same pointer and it moves by 2, it can't be forcibly aligned. */
+    if ((((size_t) dst) & 15) == 0) {
+        /* Aligned! Do SSE blocks as long as we have 16 bytes available. */
+        const __m128 divby2 = _mm_set1_ps(0.5f);
+        while (i >= 4) {   /* 4 * float32 */
+            _mm_store_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_load_ps(src), _mm_load_ps(src+4)), divby2));
+            i -= 4; src += 8; dst += 4;
+        }
+    }
+
+    /* Finish off any leftovers with scalar operations. */
+    while (i) {
+        *dst = (src[0] + src[1]) * 0.5f;
+        dst++; i--; src += 2;
+    }
+
+    cvt->len_cvt /= 2;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+#endif
+
+/* Convert from stereo to mono. Average left and right. */
+static void SDLCALL
+SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float *dst = (float *) cvt->buf;
+    const float *src = dst;
+    int i;
+
+    LOG_DEBUG_CONVERT("stereo", "mono");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    for (i = cvt->len_cvt / 8; i; --i, src += 2) {
+        *(dst++) = (src[0] + src[1]) * 0.5f;
+    }
+
+    cvt->len_cvt /= 2;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Convert from 5.1 to stereo. Average left and right, distribute center, discard LFE. */
+static void SDLCALL
+SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float *dst = (float *) cvt->buf;
+    const float *src = dst;
+    int i;
+
+    LOG_DEBUG_CONVERT("5.1", "stereo");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    /* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
+    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
+        const float front_center_distributed = src[2] * 0.5f;
+        dst[0] = (src[0] + front_center_distributed + src[4]) / 2.5f;  /* left */
+        dst[1] = (src[1] + front_center_distributed + src[5]) / 2.5f;  /* right */
+    }
+
+    cvt->len_cvt /= 3;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Convert from quad to stereo. Average left and right. */
+static void SDLCALL
+SDL_ConvertQuadToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float *dst = (float *) cvt->buf;
+    const float *src = dst;
+    int i;
+
+    LOG_DEBUG_CONVERT("quad", "stereo");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    for (i = cvt->len_cvt / (sizeof (float) * 4); i; --i, src += 4, dst += 2) {
+        dst[0] = (src[0] + src[2]) * 0.5f; /* left */
+        dst[1] = (src[1] + src[3]) * 0.5f; /* right */
+    }
+
+    cvt->len_cvt /= 2;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Convert from 7.1 to 5.1. Distribute sides across front and back. */
+static void SDLCALL
+SDL_Convert71To51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float *dst = (float *) cvt->buf;
+    const float *src = dst;
+    int i;
+
+    LOG_DEBUG_CONVERT("7.1", "5.1");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    for (i = cvt->len_cvt / (sizeof (float) * 8); i; --i, src += 8, dst += 6) {
+        const float surround_left_distributed = src[6] * 0.5f;
+        const float surround_right_distributed = src[7] * 0.5f;
+        dst[0] = (src[0] + surround_left_distributed) / 1.5f;  /* FL */
+        dst[1] = (src[1] + surround_right_distributed) / 1.5f;  /* FR */
+        dst[2] = src[2] / 1.5f; /* CC */
+        dst[3] = src[3] / 1.5f; /* LFE */
+        dst[4] = (src[4] + surround_left_distributed) / 1.5f;  /* BL */
+        dst[5] = (src[5] + surround_right_distributed) / 1.5f;  /* BR */
+    }
+
+    cvt->len_cvt /= 8;
+    cvt->len_cvt *= 6;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Convert from 5.1 to quad. Distribute center across front, discard LFE. */
+static void SDLCALL
+SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float *dst = (float *) cvt->buf;
+    const float *src = dst;
+    int i;
+
+    LOG_DEBUG_CONVERT("5.1", "quad");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    /* SDL's 4.0 layout: FL+FR+BL+BR */
+    /* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
+    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
+        const float front_center_distributed = src[2] * 0.5f;
+        dst[0] = (src[0] + front_center_distributed) / 1.5f;  /* FL */
+        dst[1] = (src[1] + front_center_distributed) / 1.5f;  /* FR */
+        dst[2] = src[4] / 1.5f;  /* BL */
+        dst[3] = src[5] / 1.5f;  /* BR */
+    }
+
+    cvt->len_cvt /= 6;
+    cvt->len_cvt *= 4;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Upmix mono to stereo (by duplication) */
+static void SDLCALL
+SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
+    float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
+    int i;
+
+    LOG_DEBUG_CONVERT("mono", "stereo");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    for (i = cvt->len_cvt / sizeof (float); i; --i) {
+        src--;
+        dst -= 2;
+        dst[0] = dst[1] = *src;
+    }
+
+    cvt->len_cvt *= 2;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Upmix stereo to a pseudo-5.1 stream */
+static void SDLCALL
+SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    int i;
+    float lf, rf, ce;
+    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
+    float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);
+
+    LOG_DEBUG_CONVERT("stereo", "5.1");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
+        dst -= 6;
+        src -= 2;
+        lf = src[0];
+        rf = src[1];
+        ce = (lf + rf) * 0.5f;
+        /* !!! FIXME: FL and FR may clip */
+        dst[0] = lf + (lf - ce);  /* FL */
+        dst[1] = rf + (rf - ce);  /* FR */
+        dst[2] = ce;  /* FC */
+        dst[3] = 0;   /* LFE (only meant for special LFE effects) */
+        dst[4] = lf;  /* BL */
+        dst[5] = rf;  /* BR */
+    }
+
+    cvt->len_cvt *= 3;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Upmix quad to a pseudo-5.1 stream */
+static void SDLCALL
+SDL_ConvertQuadTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    int i;
+    float lf, rf, lb, rb, ce;
+    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
+    float *dst = (float *) (cvt->buf + cvt->len_cvt * 3 / 2);
+
+    LOG_DEBUG_CONVERT("quad", "5.1");
+    SDL_assert(format == AUDIO_F32SYS);
+    SDL_assert(cvt->len_cvt % (sizeof(float) * 4) == 0);
+
+    for (i = cvt->len_cvt / (sizeof(float) * 4); i; --i) {
+        dst -= 6;
+        src -= 4;
+        lf = src[0];
+        rf = src[1];
+        lb = src[2];
+        rb = src[3];
+        ce = (lf + rf) * 0.5f;
+        /* !!! FIXME: FL and FR may clip */
+        dst[0] = lf + (lf - ce);  /* FL */
+        dst[1] = rf + (rf - ce);  /* FR */
+        dst[2] = ce;  /* FC */
+        dst[3] = 0;   /* LFE (only meant for special LFE effects) */
+        dst[4] = lb;  /* BL */
+        dst[5] = rb;  /* BR */
+    }
+
+    cvt->len_cvt = cvt->len_cvt * 3 / 2;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Upmix stereo to a pseudo-4.0 stream (by duplication) */
+static void SDLCALL
+SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
+    float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
+    float lf, rf;
+    int i;
+
+    LOG_DEBUG_CONVERT("stereo", "quad");
+    SDL_assert(format == AUDIO_F32SYS);
+
+    for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
+        dst -= 4;
+        src -= 2;
+        lf = src[0];
+        rf = src[1];
+        dst[0] = lf;  /* FL */
+        dst[1] = rf;  /* FR */
+        dst[2] = lf;  /* BL */
+        dst[3] = rf;  /* BR */
+    }
+
+    cvt->len_cvt *= 2;
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+
+/* Upmix 5.1 to 7.1 */
+static void SDLCALL
+SDL_Convert51To71(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+    float lf, rf, lb, rb, ls, rs;
+    int i;
+    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
+    float *dst = (float *) (cvt->buf + cvt->len_cvt * 4 / 3);
+
+    LOG_DEBUG_CONVERT("5.1", "7.1");
+    SDL_assert(format == AUDIO_F32SYS);
+    SDL_assert(cvt->len_cvt % (sizeof(float) * 6) == 0);
+
+    for (i = cvt->len_cvt / (sizeof(float) * 6); i; --i) {
+        dst -= 8;
+        src -= 6;
+        lf = src[0];
+        rf = src[1];
+        lb = src[4];
+        rb = src[5];
+        ls = (lf + lb) * 0.5f;
+        rs = (rf + rb) * 0.5f;
+        /* !!! FIXME: these four may clip */
+        lf += lf - ls;
+        rf += rf - ls;
+        lb += lb - ls;
+        rb += rb - ls;
+        dst[3] = src[3];  /* LFE */
+        dst[2] = src[2];  /* FC */
+        dst[7] = rs; /* SR */
+        dst[6] = ls; /* SL */
+        dst[5] = rb;  /* BR */
+        dst[4] = lb;  /* BL */
+        dst[1] = rf;  /* FR */
+        dst[0] = lf;  /* FL */
+    }
+
+    cvt->len_cvt = cvt->len_cvt * 4 / 3;
+
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index] (cvt, format);
+    }
+}
+
+/* SDL's resampler uses a "bandlimited interpolation" algorithm:
+     https://ccrma.stanford.edu/~jos/resample/ */
+
+#define RESAMPLER_ZERO_CROSSINGS 5
+#define RESAMPLER_BITS_PER_SAMPLE 16
+#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING  (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))
+#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1)
+
+/* This is a "modified" bessel function, so you can't use POSIX j0() */
+static double
+bessel(const double x)
+{
+    const double xdiv2 = x / 2.0;
+    double i0 = 1.0f;
+    double f = 1.0f;
+    int i = 1;
+
+    while (SDL_TRUE) {
+        const double diff = SDL_pow(xdiv2, i * 2) / SDL_pow(f, 2);
+        if (diff < 1.0e-21f) {
+            break;
+        }
+        i0 += diff;
+        i++;
+        f *= (double) i;
+    }
+
+    return i0;
+}
+
+/* build kaiser table with cardinal sine applied to it, and array of differences between elements. */
+static void
+kaiser_and_sinc(float *table, float *diffs, const int tablelen, const double beta)
+{
+    const int lenm1 = tablelen - 1;
+    const int lenm1div2 = lenm1 / 2;
+    int i;
+
+    table[0] = 1.0f;
+    for (i = 1; i < tablelen; i++) {
+        const double kaiser = bessel(beta * SDL_sqrt(1.0 - SDL_pow(((i - lenm1) / 2.0) / lenm1div2, 2.0))) / bessel(beta);
+        table[tablelen - i] = (float) kaiser;
+    }
+
+    for (i = 1; i < tablelen; i++) {
+        const float x = (((float) i) / ((float) RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) * ((float) M_PI);
+        table[i] *= SDL_sinf(x) / x;
+        diffs[i - 1] = table[i] - table[i - 1];
+    }
+    diffs[lenm1] = 0.0f;
+}
+
+
+static SDL_SpinLock ResampleFilterSpinlock = 0;
+static float *ResamplerFilter = NULL;
+static float *ResamplerFilterDifference = NULL;
+
+int
+SDL_PrepareResampleFilter(void)
+{
+    SDL_AtomicLock(&ResampleFilterSpinlock);
+    if (!ResamplerFilter) {
+        /* if dB > 50, beta=(0.1102 * (dB - 8.7)), according to Matlab. */
+        const double dB = 80.0;
+        const double beta = 0.1102 * (dB - 8.7);
+        const size_t alloclen = RESAMPLER_FILTER_SIZE * sizeof (float);
+
+        ResamplerFilter = (float *) SDL_malloc(alloclen);
+        if (!ResamplerFilter) {
+            SDL_AtomicUnlock(&ResampleFilterSpinlock);
+            return SDL_OutOfMemory();
+        }
+
+        ResamplerFilterDifference = (float *) SDL_malloc(alloclen);
+        if (!ResamplerFilterDifference) {
+            SDL_free(ResamplerFilter);
+            ResamplerFilter = NULL;
+            SDL_AtomicUnlock(&ResampleFilterSpinlock);
+            return SDL_OutOfMemory();
+        }
+        kaiser_and_sinc(ResamplerFilter, ResamplerFilterDifference, RESAMPLER_FILTER_SIZE, beta);
+    }
+    SDL_AtomicUnlock(&ResampleFilterSpinlock);
+    return 0;
+}
+
+void
+SDL_FreeResampleFilter(void)
+{
+    SDL_free(ResamplerFilter);
+    SDL_free(ResamplerFilterDifference);
+    ResamplerFilter = NULL;
+    ResamplerFilterDifference = NULL;
+}
+
+static int
+ResamplerPadding(const int inrate, const int outrate)
+{
+    if (inrate == outrate) {
+        return 0;
+    } else if (inrate > outrate) {
+        return (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
+    }
+    return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
+}
+
+/* lpadding and rpadding are expected to be buffers of (ResamplePadding(inrate, outrate) * chans * sizeof (float)) bytes. */
+static int
+SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
+                        const float *lpadding, const float *rpadding,
+                        const float *inbuf, const int inbuflen,
+                        float *outbuf, const int outbuflen)
+{
+    const double finrate = (double) inrate;
+    const double outtimeincr = 1.0 / ((float) outrate);
+    const double  ratio = ((float) outrate) / ((float) inrate);
+    const int paddinglen = ResamplerPadding(inrate, outrate);
+    const int framelen = chans * (int)sizeof (float);
+    const int inframes = inbuflen / framelen;
+    const int wantedoutframes = (int) ((inbuflen / framelen) * ratio);  /* outbuflen isn't total to write, it's total available. */
+    const int maxoutframes = outbuflen / framelen;
+    const int outframes = SDL_min(wantedoutframes, maxoutframes);
+    float *dst = outbuf;
+    double outtime = 0.0;
+    int i, j, chan;
+
+    for (i = 0; i < outframes; i++) {
+        const int srcindex = (int) (outtime * inrate);
+        const double intime = ((double) srcindex) / finrate;
+        const double innexttime = ((double) (srcindex + 1)) / finrate;
+        const double interpolation1 = 1.0 - ((innexttime - outtime) / (innexttime - intime));
+        const int filterindex1 = (int) (interpolation1 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
+        const double interpolation2 = 1.0 - interpolation1;
+        const int filterindex2 = (int) (interpolation2 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
+
+        for (chan = 0; chan < chans; chan++) {
+            float outsample = 0.0f;
+
+            /* do this twice to calculate the sample, once for the "left wing" and then same for the right. */
+            /* !!! FIXME: do both wings in one loop */
+            for (j = 0; (filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
+                const int srcframe = srcindex - j;
+                /* !!! FIXME: we can bubble this conditional out of here by doing a pre loop. */
+                const float insample = (srcframe < 0) ? lpadding[((paddinglen + srcframe) * chans) + chan] : inbuf[(srcframe * chans) + chan];
+                outsample += (float)(insample * (ResamplerFilter[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation1 * ResamplerFilterDifference[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
+            }
+
+            for (j = 0; (filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
+                const int srcframe = srcindex + 1 + j;
+                /* !!! FIXME: we can bubble this conditional out of here by doing a post loop. */
+                const float insample = (srcframe >= inframes) ? rpadding[((srcframe - inframes) * chans) + chan] : inbuf[(srcframe * chans) + chan];
+                outsample += (float)(insample * (ResamplerFilter[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation2 * ResamplerFilterDifference[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
+            }
+            *(dst++) = outsample;
+        }
+
+        outtime += outtimeincr;
+    }
+
+    return outframes * chans * sizeof (float);
+}
+
+int
+SDL_ConvertAudio(SDL_AudioCVT * cvt)
+{
+    /* !!! FIXME: (cvt) should be const; stack-copy it here. */
+    /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
+
+    /* Make sure there's data to convert */
+    if (cvt->buf == NULL) {
+        return SDL_SetError("No buffer allocated for conversion");
+    }
+
+    /* Return okay if no conversion is necessary */
+    cvt->len_cvt = cvt->len;
+    if (cvt->filters[0] == NULL) {
+        return 0;
+    }
+
+    /* Set up the conversion and go! */
+    cvt->filter_index = 0;
+    cvt->filters[0] (cvt, cvt->src_format);
+    return 0;
+}
+
+static void SDLCALL
+SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
+{
+#if DEBUG_CONVERT
+    printf("Converting byte order\n");
+#endif
+
+    switch (SDL_AUDIO_BITSIZE(format)) {
+        #define CASESWAP(b) \
+            case b: { \
+                Uint##b *ptr = (Uint##b *) cvt->buf; \
+                int i; \
+                for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
+                    *ptr = SDL_Swap##b(*ptr); \
+                } \
+                break; \
+            }
+
+        CASESWAP(16);
+        CASESWAP(32);
+        CASESWAP(64);
+
+        #undef CASESWAP
+
+        default: SDL_assert(!"unhandled byteswap datatype!"); break;
+    }
+
+    if (cvt->filters[++cvt->filter_index]) {
+        /* flip endian flag for data. */
+        if (format & SDL_AUDIO_MASK_ENDIAN) {
+            format &= ~SDL_AUDIO_MASK_ENDIAN;
+        } else {
+            format |= SDL_AUDIO_MASK_ENDIAN;
+        }
+        cvt->filters[cvt->filter_index](cvt, format);
+    }
+}
+
+static int
+SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter)
+{
+    if (cvt->filter_index >= SDL_AUDIOCVT_MAX_FILTERS) {
+        return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS);
+    }
+    if (filter == NULL) {
+        return SDL_SetError("Audio filter pointer is NULL");
+    }
+    cvt->filters[cvt->filter_index++] = filter;
+    cvt->filters[cvt->filter_index] = NULL; /* Moving terminator */
+    return 0;
+}
+
+static int
+SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
+{
+    int retval = 0;  /* 0 == no conversion necessary. */
+
+    if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
+        if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
+            return -1;
+        }
+        retval = 1;  /* added a converter. */
+    }
+
+    if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
+        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
+        const Uint16 dst_bitsize = 32;
+        SDL_AudioFilter filter = NULL;
+
+        switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
+            case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
+            case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
+            case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
+            case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
+            case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
+            default: SDL_assert(!"Unexpected audio format!"); break;
+        }
+
+        if (!filter) {
+            return SDL_SetError("No conversion from source format to float available");
+        }
+
+        if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
+            return -1;
+        }
+        if (src_bitsize < dst_bitsize) {
+            const int mult = (dst_bitsize / src_bitsize);
+            cvt->len_mult *= mult;
+            cvt->len_ratio *= mult;
+        } else if (src_bitsize > dst_bitsize) {
+            cvt->len_ratio /= (src_bitsize / dst_bitsize);
+        }
+
+        retval = 1;  /* added a converter. */
+    }
+
+    return retval;
+}
+
+static int
+SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
+{
+    int retval = 0;  /* 0 == no conversion necessary. */
+
+    if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
+        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
+        const Uint16 src_bitsize = 32;
+        SDL_AudioFilter filter = NULL;
+        switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
+            case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
+            case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
+            case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
+            case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
+            case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
+            default: SDL_assert(!"Unexpected audio format!"); break;
+        }
+
+        if (!filter) {
+            return SDL_SetError("No conversion from float to destination format available");
+        }
+
+        if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
+            return -1;
+        }
+        if (src_bitsize < dst_bitsize) {
+            const int mult = (dst_bitsize / src_bitsize);
+            cvt->len_mult *= mult;
+            cvt->len_ratio *= mult;
+        } else if (src_bitsize > dst_bitsize) {
+            cvt->len_ratio /= (src_bitsize / dst_bitsize);
+        }
+        retval = 1;  /* added a converter. */
+    }
+
+    if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
+        if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
+            return -1;
+        }
+        retval = 1;  /* added a converter. */
+    }
+
+    return retval;
+}
+
+static void
+SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
+{
+    /* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator).
+       !!! FIXME in 2.1:   We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
+       !!! FIXME in 2.1:   so we steal the ninth and tenth slot.  :( */
+    const int inrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1];
+    const int outrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS];
+    const float *src = (const float *) cvt->buf;
+    const int srclen = cvt->len_cvt;
+    /*float *dst = (float *) cvt->buf;
+    const int dstlen = (cvt->len * cvt->len_mult);*/
+    /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
+    float *dst = (float *) (cvt->buf + srclen);
+    const int dstlen = (cvt->len * cvt->len_mult) - srclen;
+    const int paddingsamples = (ResamplerPadding(inrate, outrate) * chans);
+    float *padding;
+
+    SDL_assert(format == AUDIO_F32SYS);
+
+    /* we keep no streaming state here, so pad with silence on both ends. */
+    padding = (float *) SDL_calloc(paddingsamples ? paddingsamples : 1, sizeof (float));
+    if (!padding) {
+        SDL_OutOfMemory();
+        return;
+    }
+
+    cvt->len_cvt = SDL_ResampleAudio(chans, inrate, outrate, padding, padding, src, srclen, dst, dstlen);
+
+    SDL_free(padding);
+
+    SDL_memmove(cvt->buf, dst, cvt->len_cvt);  /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
+
+    if (cvt->filters[++cvt->filter_index]) {
+        cvt->filters[cvt->filter_index](cvt, format);
+    }
+}
+
+/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
+   !!! FIXME:  store channel info, so we have to have function entry
+   !!! FIXME:  points for each supported channel count and multiple
+   !!! FIXME:  vs arbitrary. When we rev the ABI, clean this up. */
+#define RESAMPLER_FUNCS(chans) \
+    static void SDLCALL \
+    SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
+        SDL_ResampleCVT(cvt, chans, format); \
+    }
+RESAMPLER_FUNCS(1)
+RESAMPLER_FUNCS(2)
+RESAMPLER_FUNCS(4)
+RESAMPLER_FUNCS(6)
+RESAMPLER_FUNCS(8)
+#undef RESAMPLER_FUNCS
+
+static SDL_AudioFilter
+ChooseCVTResampler(const int dst_channels)
+{
+    switch (dst_channels) {
+        case 1: return SDL_ResampleCVT_c1;
+        case 2: return SDL_ResampleCVT_c2;
+        case 4: return SDL_ResampleCVT_c4;
+        case 6: return SDL_ResampleCVT_c6;
+        case 8: return SDL_ResampleCVT_c8;
+        default: break;
+    }
+
+    return NULL;
+}
+
+static int
+SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
+                          const int src_rate, const int dst_rate)
+{
+    SDL_AudioFilter filter;
+
+    if (src_rate == dst_rate) {
+        return 0;  /* no conversion necessary. */
+    }
+
+    filter = ChooseCVTResampler(dst_channels);
+    if (filter == NULL) {
+        return SDL_SetError("No conversion available for these rates");
+    }
+
+    if (SDL_PrepareResampleFilter() < 0) {
+        return -1;
+    }
+
+    /* Update (cvt) with filter details... */
+    if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
+        return -1;
+    }
+
+    /* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator).
+       !!! FIXME in 2.1:   We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
+       !!! FIXME in 2.1:   so we steal the ninth and tenth slot.  :( */
+    if (cvt->filter_index >= (SDL_AUDIOCVT_MAX_FILTERS-2)) {
+        return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS-2);
+    }
+    cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1] = (SDL_AudioFilter) (size_t) src_rate;
+    cvt->filters[SDL_AUDIOCVT_MAX_FILTERS] = (SDL_AudioFilter) (size_t) dst_rate;
+
+    if (src_rate < dst_rate) {
+        const double mult = ((double) dst_rate) / ((double) src_rate);
+        cvt->len_mult *= (int) SDL_ceil(mult);
+        cvt->len_ratio *= mult;
+    } else {
+        cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
+    }
+
+    /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
+    /* the buffer is big enough to hold the destination now, but
+       we need it large enough to hold a separate scratch buffer. */
+    cvt->len_mult *= 2;
+
+    return 1;               /* added a converter. */
+}
+
+static SDL_bool
+SDL_SupportedAudioFormat(const SDL_AudioFormat fmt)
+{
+    switch (fmt) {
+        case AUDIO_U8:
+        case AUDIO_S8:
+        case AUDIO_U16LSB:
+        case AUDIO_S16LSB:
+        case AUDIO_U16MSB:
+        case AUDIO_S16MSB:
+        case AUDIO_S32LSB:
+        case AUDIO_S32MSB:
+        case AUDIO_F32LSB:
+        case AUDIO_F32MSB:
+            return SDL_TRUE;  /* supported. */
+
+        default:
+            break;
+    }
+
+    return SDL_FALSE;  /* unsupported. */
+}
+
+static SDL_bool
+SDL_SupportedChannelCount(const int channels)
+{
+    switch (channels) {
+        case 1:  /* mono */
+        case 2:  /* stereo */
+        case 4:  /* quad */
+        case 6:  /* 5.1 */
+        case 8:  /* 7.1 */
+          return SDL_TRUE;  /* supported. */
+
+        default:
+            break;
+    }
+
+    return SDL_FALSE;  /* unsupported. */
+}
+
+
+/* Creates a set of audio filters to convert from one format to another.
+   Returns 0 if no conversion is needed, 1 if the audio filter is set up,
+   or -1 if an error like invalid parameter, unsupported format, etc. occurred.
+*/
+
+int
+SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
+                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
+                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
+{
+    /* Sanity check target pointer */
+    if (cvt == NULL) {
+        return SDL_InvalidParamError("cvt");
+    }
+
+    /* Make sure we zero out the audio conversion before error checking */
+    SDL_zerop(cvt);
+
+    if (!SDL_SupportedAudioFormat(src_fmt)) {
+        return SDL_SetError("Invalid source format");
+    } else if (!SDL_SupportedAudioFormat(dst_fmt)) {
+        return SDL_SetError("Invalid destination format");
+    } else if (!SDL_SupportedChannelCount(src_channels)) {
+        return SDL_SetError("Invalid source channels");
+    } else if (!SDL_SupportedChannelCount(dst_channels)) {
+        return SDL_SetError("Invalid destination channels");
+    } else if (src_rate == 0) {
+        return SDL_SetError("Source rate is zero");
+    } else if (dst_rate == 0) {
+        return SDL_SetError("Destination rate is zero");
+    }
+
+#if DEBUG_CONVERT
+    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
+           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
+#endif
+
+    /* Start off with no conversion necessary */
+    cvt->src_format = src_fmt;
+    cvt->dst_format = dst_fmt;
+    cvt->needed = 0;
+    cvt->filter_index = 0;
+    SDL_zero(cvt->filters);
+    cvt->len_mult = 1;
+    cvt->len_ratio = 1.0;
+    cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
+
+    /* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
+    SDL_ChooseAudioConverters();
+
+    /* Type conversion goes like this now:
+        - byteswap to CPU native format first if necessary.
+        - convert to native Float32 if necessary.
+        - resample and change channel count if necessary.
+        - convert back to native format.
+        - byteswap back to foreign format if necessary.
+
+       The expectation is we can process data faster in float32
+       (possibly with SIMD), and making several passes over the same
+       buffer is likely to be CPU cache-friendly, avoiding the
+       biggest performance hit in modern times. Previously we had
+       (script-generated) custom converters for every data type and
+       it was a bloat on SDL compile times and final library size. */
+
+    /* see if we can skip float conversion entirely. */
+    if (src_rate == dst_rate && src_channels == dst_channels) {
+        if (src_fmt == dst_fmt) {
+            return 0;
+        }
+
+        /* just a byteswap needed? */
+        if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
+                return -1;
+            }
+            cvt->needed = 1;
+            return 1;
+        }
+    }
+
+    /* Convert data types, if necessary. Updates (cvt). */
+    if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
+        return -1;              /* shouldn't happen, but just in case... */
+    }
+
+    /* Channel conversion */
+    if (src_channels < dst_channels) {
+        /* Upmixing */
+        /* Mono -> Stereo [-> ...] */
+        if ((src_channels == 1) && (dst_channels > 1)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
+                return -1;
+            }
+            cvt->len_mult *= 2;
+            src_channels = 2;
+            cvt->len_ratio *= 2;
+        }
+        /* [Mono ->] Stereo -> 5.1 [-> 7.1] */
+        if ((src_channels == 2) && (dst_channels >= 6)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) {
+                return -1;
+            }
+            src_channels = 6;
+            cvt->len_mult *= 3;
+            cvt->len_ratio *= 3;
+        }
+        /* Quad -> 5.1 [-> 7.1] */
+        if ((src_channels == 4) && (dst_channels >= 6)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadTo51) < 0) {
+                return -1;
+            }
+            src_channels = 6;
+            cvt->len_mult = (cvt->len_mult * 3 + 1) / 2;
+            cvt->len_ratio *= 1.5;
+        }
+        /* [[Mono ->] Stereo ->] 5.1 -> 7.1 */
+        if ((src_channels == 6) && (dst_channels == 8)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51To71) < 0) {
+                return -1;
+            }
+            src_channels = 8;
+            cvt->len_mult = (cvt->len_mult * 4 + 2) / 3;
+            /* Should be numerically exact with every valid input to this
+               function */
+            cvt->len_ratio = cvt->len_ratio * 4 / 3;
+        }
+        /* [Mono ->] Stereo -> Quad */
+        if ((src_channels == 2) && (dst_channels == 4)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoToQuad) < 0) {
+                return -1;
+            }
+            src_channels = 4;
+            cvt->len_mult *= 2;
+            cvt->len_ratio *= 2;
+        }
+    } else if (src_channels > dst_channels) {
+        /* Downmixing */
+        /* 7.1 -> 5.1 [-> Stereo [-> Mono]] */
+        /* 7.1 -> 5.1 [-> Quad] */
+        if ((src_channels == 8) && (dst_channels <= 6)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert71To51) < 0) {
+                return -1;
+            }
+            src_channels = 6;
+            cvt->len_ratio *= 0.75;
+        }
+        /* [7.1 ->] 5.1 -> Stereo [-> Mono] */
+        if ((src_channels == 6) && (dst_channels <= 2)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToStereo) < 0) {
+                return -1;
+            }
+            src_channels = 2;
+            cvt->len_ratio /= 3;
+        }
+        /* 5.1 -> Quad */
+        if ((src_channels == 6) && (dst_channels == 4)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToQuad) < 0) {
+                return -1;
+            }
+            src_channels = 4;
+            cvt->len_ratio = cvt->len_ratio * 2 / 3;
+        }
+        /* Quad -> Stereo [-> Mono] */
+        if ((src_channels == 4) && (dst_channels <= 2)) {
+            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadToStereo) < 0) {
+                return -1;
+            }
+            src_channels = 2;
+            cvt->len_ratio /= 2;
+        }
+        /* [... ->] Stereo -> Mono */
+        if ((src_channels == 2) && (dst_channels == 1)) {
+            SDL_AudioFilter filter = NULL;
+
+            #if HAVE_SSE3_INTRINSICS
+            if (SDL_HasSSE3()) {
+                filter = SDL_ConvertStereoToMono_SSE3;
+            }
+            #endif
+
+            if (!filter) {
+                filter = SDL_ConvertStereoToMono;
+            }
+
+            if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
+                return -1;
+            }
+
+            src_channels = 1;
+            cvt->len_ratio /= 2;
+        }
+    }
+
+    if (src_channels != dst_channels) {
+        /* All combinations of supported channel counts should have been
+           handled by now, but let's be defensive */
+      return SDL_SetError("Invalid channel combination");
+    }
+    
+    /* Do rate conversion, if necessary. Updates (cvt). */
+    if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
+        return -1;              /* shouldn't happen, but just in case... */
+    }
+
+    /* Move to final data type. */
+    if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
+        return -1;              /* shouldn't happen, but just in case... */
+    }
+
+    cvt->needed = (cvt->filter_index != 0);
+    return (cvt->needed);
+}
+
+typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *inbuf, const int inbuflen, void *outbuf, const int outbuflen);
+typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
+typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
+
+struct _SDL_AudioStream
+{
+    SDL_AudioCVT cvt_before_resampling;
+    SDL_AudioCVT cvt_after_resampling;
+    SDL_DataQueue *queue;
+    SDL_bool first_run;
+    Uint8 *staging_buffer;
+    int staging_buffer_size;
+    int staging_buffer_filled;
+    Uint8 *work_buffer_base;  /* maybe unaligned pointer from SDL_realloc(). */
+    int work_buffer_len;
+    int src_sample_frame_size;
+    SDL_AudioFormat src_format;
+    Uint8 src_channels;
+    int src_rate;
+    int dst_sample_frame_size;
+    SDL_AudioFormat dst_format;
+    Uint8 dst_channels;
+    int dst_rate;
+    double rate_incr;
+    Uint8 pre_resample_channels;
+    int packetlen;
+    int resampler_padding_samples;
+    float *resampler_padding;
+    void *resampler_state;
+    SDL_ResampleAudioStreamFunc resampler_func;
+    SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
+    SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
+};
+
+static Uint8 *
+EnsureStreamBufferSize(SDL_AudioStream *stream, const int newlen)
+{
+    Uint8 *ptr;
+    size_t offset;
+
+    if (stream->work_buffer_len >= newlen) {
+        ptr = stream->work_buffer_base;
+    } else {
+        ptr = (Uint8 *) SDL_realloc(stream->work_buffer_base, newlen + 32);
+        if (!ptr) {
+            SDL_OutOfMemory();
+            return NULL;
+        }
+        /* Make sure we're aligned to 16 bytes for SIMD code. */
+        stream->work_buffer_base = ptr;
+        stream->work_buffer_len = newlen;
+    }
+
+    offset = ((size_t) ptr) & 15;
+    return offset ? ptr + (16 - offset) : ptr;
+}
+
+#ifdef HAVE_LIBSAMPLERATE_H
+static int
+SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
+{
+    const float *inbuf = (const float *) _inbuf;
+    float *outbuf = (float *) _outbuf;
+    const int framelen = sizeof(float) * stream->pre_resample_channels;
+    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
+    SRC_DATA data;
+    int result;
+
+    SDL_assert(inbuf != ((const float *) outbuf));  /* SDL_AudioStreamPut() shouldn't allow in-place resamples. */
+
+    data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */
+    data.input_frames = inbuflen / framelen;
+    data.input_frames_used = 0;
+
+    data.data_out = outbuf;
+    data.output_frames = outbuflen / framelen;
+
+    data.end_of_input = 0;
+    data.src_ratio = stream->rate_incr;
+
+    result = SRC_src_process(state, &data);
+    if (result != 0) {
+        SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
+        return 0;
+    }
+
+    /* If this fails, we need to store them off somewhere */
+    SDL_assert(data.input_frames_used == data.input_frames);
+
+    return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
+}
+
+static void
+SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
+{
+    SRC_src_reset((SRC_STATE *)stream->resampler_state);
+}
+
+static void
+SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
+{
+    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
+    if (state) {
+        SRC_src_delete(state);
+    }
+
+    stream->resampler_state = NULL;
+    stream->resampler_func = NULL;
+    stream->reset_resampler_func = NULL;
+    stream->cleanup_resampler_func = NULL;
+}
+
+static SDL_bool
+SetupLibSampleRateResampling(SDL_AudioStream *stream)
+{
+    int result = 0;
+    SRC_STATE *state = NULL;
+
+    if (SRC_available) {
+        state = SRC_src_new(SRC_converter, stream->pre_resample_channels, &result);
+        if (!state) {
+            SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
+        }
+    }
+
+    if (!state) {
+        SDL_CleanupAudioStreamResampler_SRC(stream);
+        return SDL_FALSE;
+    }
+
+    stream->resampler_state = state;
+    stream->resampler_func = SDL_ResampleAudioStream_SRC;
+    stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
+    stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
+
+    return SDL_TRUE;
+}
+#endif /* HAVE_LIBSAMPLERATE_H */
+
+
+static int
+SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
+{
+    const Uint8 *inbufend = ((const Uint8 *) _inbuf) + inbuflen;
+    const float *inbuf = (const float *) _inbuf;
+    float *outbuf = (float *) _outbuf;
+    const int chans = (int) stream->pre_resample_channels;
+    const int inrate = stream->src_rate;
+    const int outrate = stream->dst_rate;
+    const int paddingsamples = stream->resampler_padding_samples;
+    const int paddingbytes = paddingsamples * sizeof (float);
+    float *lpadding = (float *) stream->resampler_state;
+    const float *rpadding = (const float *) inbufend; /* we set this up so there are valid padding samples at the end of the input buffer. */
+    const int cpy = SDL_min(inbuflen, paddingbytes);
+    int retval;
+
+    SDL_assert(inbuf != ((const float *) outbuf));  /* SDL_AudioStreamPut() shouldn't allow in-place resamples. */
+
+    retval = SDL_ResampleAudio(chans, inrate, outrate, lpadding, rpadding, inbuf, inbuflen, outbuf, outbuflen);
+
+    /* update our left padding with end of current input, for next run. */
+    SDL_memcpy((lpadding + paddingsamples) - (cpy / sizeof (float)), inbufend - cpy, cpy);
+    return retval;
+}
+
+static void
+SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
+{
+    /* set all the padding to silence. */
+    const int len = stream->resampler_padding_samples;
+    SDL_memset(stream->resampler_state, '\0', len * sizeof (float));
+}
+
+static void
+SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
+{
+    SDL_free(stream->resampler_state);
+}
+
+SDL_AudioStream *
+SDL_NewAudioStream(const SDL_AudioFormat src_format,
+                   const Uint8 src_channels,
+                   const int src_rate,
+                   const SDL_AudioFormat dst_format,
+                   const Uint8 dst_channels,
+                   const int dst_rate)
+{
+    const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
+    Uint8 pre_resample_channels;
+    SDL_AudioStream *retval;
+
+    retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
+    if (!retval) {
+        return NULL;
+    }
+
+    /* If increasing channels, do it after resampling, since we'd just
+       do more work to resample duplicate channels. If we're decreasing, do
+       it first so we resample the interpolated data instead of interpolating
+       the resampled data (!!! FIXME: decide if that works in practice, though!). */
+    pre_resample_channels = SDL_min(src_channels, dst_channels);
+
+    retval->first_run = SDL_TRUE;
+    retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
+    retval->src_format = src_format;
+    retval->src_channels = src_channels;
+    retval->src_rate = src_rate;
+    retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
+    retval->dst_format = dst_format;
+    retval->dst_channels = dst_channels;
+    retval->dst_rate = dst_rate;
+    retval->pre_resample_channels = pre_resample_channels;
+    retval->packetlen = packetlen;
+    retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
+    retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels;
+    retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples ? retval->resampler_padding_samples : 1, sizeof (float));
+
+    if (retval->resampler_padding == NULL) {
+        SDL_FreeAudioStream(retval);
+        SDL_OutOfMemory();
+        return NULL;
+    }
+
+    retval->staging_buffer_size = ((retval->resampler_padding_samples / retval->pre_resample_channels) * retval->src_sample_frame_size);
+    if (retval->staging_buffer_size > 0) {
+        retval->staging_buffer = (Uint8 *) SDL_malloc(retval->staging_buffer_size);
+        if (retval->staging_buffer == NULL) {
+            SDL_FreeAudioStream(retval);
+            SDL_OutOfMemory();
+            return NULL;
+        }
+    }
+
+    /* Not resampling? It's an easy conversion (and maybe not even that!) */
+    if (src_rate == dst_rate) {
+        retval->cvt_before_resampling.needed = SDL_FALSE;
+        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
+            SDL_FreeAudioStream(retval);
+            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
+        }
+    } else {
+        /* Don't resample at first. Just get us to Float32 format. */
+        /* !!! FIXME: convert to int32 on devices without hardware float. */
+        if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
+            SDL_FreeAudioStream(retval);
+            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
+        }
+
+#ifdef HAVE_LIBSAMPLERATE_H
+        SetupLibSampleRateResampling(retval);
+#endif
+
+        if (!retval->resampler_func) {
+            retval->resampler_state = SDL_calloc(retval->resampler_padding_samples, sizeof (float));
+            if (!retval->resampler_state) {
+                SDL_FreeAudioStream(retval);
+                SDL_OutOfMemory();
+                return NULL;
+            }
+
+            if (SDL_PrepareResampleFilter() < 0) {
+                SDL_free(retval->resampler_state);
+                retval->resampler_state = NULL;
+                SDL_FreeAudioStream(retval);
+                return NULL;
+            }
+
+            retval->resampler_func = SDL_ResampleAudioStream;
+            retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
+            retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
+        }
+
+        /* Convert us to the final format after resampling. */
+        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
+            SDL_FreeAudioStream(retval);
+            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
+        }
+    }
+
+    retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
+    if (!retval->queue) {
+        SDL_FreeAudioStream(retval);
+        return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
+    }
+
+    return retval;
+}
+
+static int
+SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, int *maxputbytes)
+{
+    int buflen = len;
+    int workbuflen;
+    Uint8 *workbuf;
+    Uint8 *resamplebuf = NULL;
+    int resamplebuflen = 0;
+    int neededpaddingbytes;
+    int paddingbytes;
+
+    /* !!! FIXME: several converters can take advantage of SIMD, but only
+       !!! FIXME:  if the data is aligned to 16 bytes. EnsureStreamBufferSize()
+       !!! FIXME:  guarantees the buffer will align, but the
+       !!! FIXME:  converters will iterate over the data backwards if
+       !!! FIXME:  the output grows, and this means we won't align if buflen
+       !!! FIXME:  isn't a multiple of 16. In these cases, we should chop off
+       !!! FIXME:  a few samples at the end and convert them separately. */
+
+    /* no padding prepended on first run. */
+    neededpaddingbytes = stream->resampler_padding_samples * sizeof (float);
+    paddingbytes = stream->first_run ? 0 : neededpaddingbytes;
+    stream->first_run = SDL_FALSE;
+
+    /* Make sure the work buffer can hold all the data we need at once... */
+    workbuflen = buflen;
+    if (stream->cvt_before_resampling.needed) {
+        workbuflen *= stream->cvt_before_resampling.len_mult;
+    }
+
+    if (stream->dst_rate != stream->src_rate) {
+        /* resamples can't happen in place, so make space for second buf. */
+        const int framesize = stream->pre_resample_channels * sizeof (float);
+        const int frames = workbuflen / framesize;
+        resamplebuflen = ((int) SDL_ceil(frames * stream->rate_incr)) * framesize;
+        #if DEBUG_AUDIOSTREAM
+        printf("AUDIOSTREAM: will resample %d bytes to %d (ratio=%.6f)\n", workbuflen, resamplebuflen, stream->rate_incr);
+        #endif
+        workbuflen += resamplebuflen;
+    }
+
+    if (stream->cvt_after_resampling.needed) {
+        /* !!! FIXME: buffer might be big enough already? */
+        workbuflen *= stream->cvt_after_resampling.len_mult;
+    }
+
+    workbuflen += neededpaddingbytes;
+
+    #if DEBUG_AUDIOSTREAM
+    printf("AUDIOSTREAM: Putting %d bytes of preconverted audio, need %d byte work buffer\n", buflen, workbuflen);
+    #endif
+
+    workbuf = EnsureStreamBufferSize(stream, workbuflen);
+    if (!workbuf) {
+        return -1;  /* probably out of memory. */
+    }
+
+    resamplebuf = workbuf;  /* default if not resampling. */
+
+    SDL_memcpy(workbuf + paddingbytes, buf, buflen);
+
+    if (stream->cvt_before_resampling.needed) {
+        stream->cvt_before_resampling.buf = workbuf + paddingbytes;
+        stream->cvt_before_resampling.len = buflen;
+        if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
+            return -1;   /* uhoh! */
+        }
+        buflen = stream->cvt_before_resampling.len_cvt;
+
+        #if DEBUG_AUDIOSTREAM
+        printf("AUDIOSTREAM: After initial conversion we have %d bytes\n", buflen);
+        #endif
+    }
+
+    if (stream->dst_rate != stream->src_rate) {
+        /* save off some samples at the end; they are used for padding now so
+           the resampler is coherent and then used at the start of the next
+           put operation. Prepend last put operation's padding, too. */
+
+        /* prepend prior put's padding. :P */
+        if (paddingbytes) {
+            SDL_memcpy(workbuf, stream->resampler_padding, paddingbytes);
+            buflen += paddingbytes;
+        }
+
+        /* save off the data at the end for the next run. */
+        SDL_memcpy(stream->resampler_padding, workbuf + (buflen - neededpaddingbytes), neededpaddingbytes);
+
+        resamplebuf = workbuf + buflen;  /* skip to second piece of workbuf. */
+        SDL_assert(buflen >= neededpaddingbytes);
+        if (buflen > neededpaddingbytes) {
+            buflen = stream->resampler_func(stream, workbuf, buflen - neededpaddingbytes, resamplebuf, resamplebuflen);
+        } else {
+            buflen = 0;
+        }
+
+        #if DEBUG_AUDIOSTREAM
+        printf("AUDIOSTREAM: After resampling we have %d bytes\n", buflen);
+        #endif
+    }
+
+    if (stream->cvt_after_resampling.needed && (buflen > 0)) {
+        stream->cvt_after_resampling.buf = resamplebuf;
+        stream->cvt_after_resampling.len = buflen;
+        if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
+            return -1;   /* uhoh! */
+        }
+        buflen = stream->cvt_after_resampling.len_cvt;
+
+        #if DEBUG_AUDIOSTREAM
+        printf("AUDIOSTREAM: After final conversion we have %d bytes\n", buflen);
+        #endif
+    }
+
+    #if DEBUG_AUDIOSTREAM
+    printf("AUDIOSTREAM: Final output is %d bytes\n", buflen);
+    #endif
+
+    if (maxputbytes) {
+        const int maxbytes = *maxputbytes;
+        if (buflen > maxbytes)
+            buflen = maxbytes;
+        *maxputbytes -= buflen;
+    }
+
+    /* resamplebuf holds the final output, even if we didn't resample. */
+    return buflen ? SDL_WriteToDataQueue(stream->queue, resamplebuf, buflen) : 0;
+}
+
+int
+SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
+{
+    /* !!! FIXME: several converters can take advantage of SIMD, but only
+       !!! FIXME:  if the data is aligned to 16 bytes. EnsureStreamBufferSize()
+       !!! FIXME:  guarantees the buffer will align, but the
+       !!! FIXME:  converters will iterate over the data backwards if
+       !!! FIXME:  the output grows, and this means we won't align if buflen
+       !!! FIXME:  isn't a multiple of 16. In these cases, we should chop off
+       !!! FIXME:  a few samples at the end and convert them separately. */
+
+    #if DEBUG_AUDIOSTREAM
+    printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
+    #endif
+
+    if (!stream) {
+        return SDL_InvalidParamError("stream");
+    } else if (!buf) {
+        return SDL_InvalidParamError("buf");
+    } else if (len == 0) {
+        return 0;  /* nothing to do. */
+    } else if ((len % stream->src_sample_frame_size) != 0) {
+        return SDL_SetError("Can't add partial sample frames");
+    }
+
+    if (!stream->cvt_before_resampling.needed &&
+        (stream->dst_rate == stream->src_rate) &&
+        !stream->cvt_after_resampling.needed) {
+        #if DEBUG_AUDIOSTREAM
+        printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", len);
+        #endif
+        return SDL_WriteToDataQueue(stream->queue, buf, len);
+    }
+
+    while (len > 0) {
+        int amount;
+
+        /* If we don't have a staging buffer or we're given enough data that
+           we don't need to store it for later, skip the staging process.
+         */
+        if (!stream->staging_buffer_filled && len >= stream->staging_buffer_size) {
+            return SDL_AudioStreamPutInternal(stream, buf, len, NULL);
+        }
+
+        /* If there's not enough data to fill the staging buffer, just save it */
+        if ((stream->staging_buffer_filled + len) < stream->staging_buffer_size) {
+            SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, len);
+            stream->staging_buffer_filled += len;
+            return 0;
+        }
+ 
+        /* Fill the staging buffer, process it, and continue */
+        amount = (stream->staging_buffer_size - stream->staging_buffer_filled);
+        SDL_assert(amount > 0);
+        SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, amount);
+        stream->staging_buffer_filled = 0;
+        if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, NULL) < 0) {
+            return -1;
+        }
+        buf = (void *)((Uint8 *)buf + amount);
+        len -= amount;
+    }
+    return 0;
+}
+
+int SDL_AudioStreamFlush(SDL_AudioStream *stream)
+{
+    if (!stream) {
+        return SDL_InvalidParamError("stream");
+    }
+
+    #if DEBUG_AUDIOSTREAM
+    printf("AUDIOSTREAM: flushing! staging_buffer_filled=%d bytes\n", stream->staging_buffer_filled);
+    #endif
+
+    /* shouldn't use a staging buffer if we're not resampling. */
+    SDL_assert((stream->dst_rate != stream->src_rate) || (stream->staging_buffer_filled == 0));
+
+    if (stream->staging_buffer_filled > 0) {
+        /* push the staging buffer + silence. We need to flush out not just
+           the staging buffer, but the piece that the stream was saving off
+           for right-side resampler padding. */
+        const SDL_bool first_run = stream->first_run;
+        const int filled = stream->staging_buffer_filled;
+        int actual_input_frames = filled / stream->src_sample_frame_size;
+        if (!first_run)
+            actual_input_frames += stream->resampler_padding_samples / stream->pre_resample_channels;
+
+        if (actual_input_frames > 0) {  /* don't bother if nothing to flush. */
+            /* This is how many bytes we're expecting without silence appended. */
+            int flush_remaining = ((int) SDL_ceil(actual_input_frames * stream->rate_incr)) * stream->dst_sample_frame_size;
+
+            #if DEBUG_AUDIOSTREAM
+            printf("AUDIOSTREAM: flushing with padding to get max %d bytes!\n", flush_remaining);
+            #endif
+
+            SDL_memset(stream->staging_buffer + filled, '\0', stream->staging_buffer_size - filled);
+            if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
+                return -1;
+            }
+
+            /* we have flushed out (or initially filled) the pending right-side
+               resampler padding, but we need to push more silence to guarantee
+               the staging buffer is fully flushed out, too. */
+            SDL_memset(stream->staging_buffer, '\0', filled);
+            if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
+                return -1;
+            }
+        }
+    }
+
+    stream->staging_buffer_filled = 0;
+    stream->first_run = SDL_TRUE;
+
+    return 0;
+}
+
+/* get converted/resampled data from the stream */
+int
+SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len)
+{
+    #if DEBUG_AUDIOSTREAM
+    printf("AUDIOSTREAM: want to get %d converted bytes\n", len);
+    #endif
+
+    if (!stream) {
+        return SDL_InvalidParamError("stream");
+    } else if (!buf) {
+        return SDL_InvalidParamError("buf");
+    } else if (len <= 0) {
+        return 0;  /* nothing to do. */
+    } else if ((len % stream->dst_sample_frame_size) != 0) {
+        return SDL_SetError("Can't request partial sample frames");
+    }
+
+    return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
+}
+
+/* number of converted/resampled bytes available */
+int
+SDL_AudioStreamAvailable(SDL_AudioStream *stream)
+{
+    return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
+}
+
+void
+SDL_AudioStreamClear(SDL_AudioStream *stream)
+{
+    if (!stream) {
+        SDL_InvalidParamError("stream");
+    } else {
+        SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
+        if (stream->reset_resampler_func) {
+            stream->reset_resampler_func(stream);
+        }
+        stream->first_run = SDL_TRUE;
+        stream->staging_buffer_filled = 0;
+    }
+}
+
+/* dispose of a stream */
+void
+SDL_FreeAudioStream(SDL_AudioStream *stream)
+{
+    if (stream) {
+        if (stream->cleanup_resampler_func) {
+            stream->cleanup_resampler_func(stream);
+        }
+        SDL_FreeDataQueue(stream->queue);
+        SDL_free(stream->staging_buffer);
+        SDL_free(stream->work_buffer_base);
+        SDL_free(stream->resampler_padding);
+        SDL_free(stream);
+    }
+}
+
+/* vi: set ts=4 sw=4 expandtab: */
+