audio: Prebake the resampler's kaiser table instead of doing it at runtime.

build-scripts/gen_audio_resampler_filter.c

@@ -0,0 +1,162 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2022 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.
+*/
+
+/*
+
+Built with:
+
+gcc -o genfilter build-scripts/gen_audio_resampler_filter.c -lm && ./genfilter > src/audio/SDL_audio_resampler_filter.h
+
+ */
+
+/*
+   SDL's resampler uses a "bandlimited interpolation" algorithm:
+     https://ccrma.stanford.edu/~jos/resample/
+
+   This code pre-generates the kaiser tables so we don't have to do this at
+   run time, at a cost of about 20 kilobytes of static data in SDL. This code
+   used to be part of SDL itself and generated the tables on the first use,
+   but that was expensive to produce on platforms without floating point
+   hardware.
+*/
+
+#include <stdio.h>
+#include <math.h>
+
+#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 (1) {
+        const double diff = pow(xdiv2, i * 2) / 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 * sqrt(1.0 - 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] *= sinf(x) / x;
+        diffs[i - 1] = table[i] - table[i - 1];
+    }
+    diffs[lenm1] = 0.0f;
+}
+
+
+static float ResamplerFilter[RESAMPLER_FILTER_SIZE];
+static float ResamplerFilterDifference[RESAMPLER_FILTER_SIZE];
+
+static void
+PrepareResampleFilter(void)
+{
+    /* 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);
+    kaiser_and_sinc(ResamplerFilter, ResamplerFilterDifference, RESAMPLER_FILTER_SIZE, beta);
+}
+
+int main(void)
+{
+    int i;
+
+    PrepareResampleFilter();
+
+    printf(
+        "/*\n"
+        "  Simple DirectMedia Layer\n"
+        "  Copyright (C) 1997-2022 Sam Lantinga <slouken@libsdl.org>\n"
+        "\n"
+        "  This software is provided 'as-is', without any express or implied\n"
+        "  warranty.  In no event will the authors be held liable for any damages\n"
+        "  arising from the use of this software.\n"
+        "\n"
+        "  Permission is granted to anyone to use this software for any purpose,\n"
+        "  including commercial applications, and to alter it and redistribute it\n"
+        "  freely, subject to the following restrictions:\n"
+        "\n"
+        "  1. The origin of this software must not be misrepresented; you must not\n"
+        "     claim that you wrote the original software. If you use this software\n"
+        "     in a product, an acknowledgment in the product documentation would be\n"
+        "     appreciated but is not required.\n"
+        "  2. Altered source versions must be plainly marked as such, and must not be\n"
+        "     misrepresented as being the original software.\n"
+        "  3. This notice may not be removed or altered from any source distribution.\n"
+        "*/\n"
+        "\n"
+        "/* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c */\n"
+        "\n"
+        "#define RESAMPLER_ZERO_CROSSINGS %d\n"
+        "#define RESAMPLER_BITS_PER_SAMPLE %d\n"
+        "#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))\n"
+        "#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1)\n"
+        "\n", RESAMPLER_ZERO_CROSSINGS, RESAMPLER_BITS_PER_SAMPLE
+    );
+
+    printf("static const float ResamplerFilter[RESAMPLER_FILTER_SIZE] = {\n");
+    printf("    %ff", ResamplerFilter[0]);
+    for (i = 0; i < RESAMPLER_FILTER_SIZE-1; i++) {
+        printf("%s%ff", ((i % 6) == 5) ? ",\n    " : ", ", ResamplerFilter[i+1]);
+    }
+    printf("\n};\n\n");
+
+    printf("static const float ResamplerFilterDifference[RESAMPLER_FILTER_SIZE] = {\n");
+    printf("    %ff", ResamplerFilterDifference[0]);
+    for (i = 0; i < RESAMPLER_FILTER_SIZE-1; i++) {
+        printf("%s%ff", ((i % 6) == 5) ? ",\n    " : ", ", ResamplerFilterDifference[i+1]);
+    }
+    printf("\n};\n\n");
+    printf("/* vi: set ts=4 sw=4 expandtab: */\n\n");
+
+    return 0;
+}
+
+/* vi: set ts=4 sw=4 expandtab: */
+

src/audio/SDL_audio.c

@@ -1649,8 +1649,6 @@ SDL_AudioQuit(void)
 #ifdef HAVE_LIBSAMPLERATE_H
     UnloadLibSampleRate();
 #endif
-
-    SDL_FreeResampleFilter();
 }
 
 #define NUM_FORMATS 10

src/audio/SDL_audio_c.h

@@ -70,11 +70,6 @@ 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: */

src/audio/SDL_audiocvt.c

@@ -704,97 +704,7 @@ SDL_Convert51To71(SDL_AudioCVT * cvt, SDL_AudioFormat format)
 /* SDL's resampler uses a "bandlimited interpolation" algorithm:
      https://ccrma.stanford.edu/~jos/resample/ */
 
-#define RESAMPLER_ZERO_CROSSINGS 5
+#include "SDL_audio_resampler_filter.h"
-#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)
@@ -803,7 +713,7 @@ ResamplerPadding(const int inrate, const int outrate)
         return 0;
     }
     if (inrate > outrate) {
-        return (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
+        return (int) SDL_ceilf(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
     }
     return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
 }
@@ -1119,10 +1029,6 @@ SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
         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;
@@ -1743,13 +1649,6 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
                 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;