Rewrote audio resampler using cubic filter interpolation

This allows using a much smaller (1.5 KB) lookup table, in exchange for a small amount of extra work per frame.

The extra work (a few extra loads/mul/adds) is negligible, and can execute in parallel.
The reduction in cache misses almost certainly outweighs any added cost.

The table is generated at runtime, and takes less than 0.02ms on my computer.

Xcode/SDL/SDL.xcodeproj/project.pbxproj

@@ -391,7 +391,6 @@
 		F32DDACF2AB795A30041EAA5 /* SDL_audio_channel_converters.h in Headers */ = {isa = PBXBuildFile; fileRef = F32DDAC92AB795A30041EAA5 /* SDL_audio_channel_converters.h */; };
 		F32DDAD02AB795A30041EAA5 /* SDL_audioresample.h in Headers */ = {isa = PBXBuildFile; fileRef = F32DDACA2AB795A30041EAA5 /* SDL_audioresample.h */; };
 		F32DDAD12AB795A30041EAA5 /* SDL_audioqueue.c in Sources */ = {isa = PBXBuildFile; fileRef = F32DDACB2AB795A30041EAA5 /* SDL_audioqueue.c */; };
-		F32DDAD22AB795A30041EAA5 /* SDL_audio_resampler_filter.h in Headers */ = {isa = PBXBuildFile; fileRef = F32DDACC2AB795A30041EAA5 /* SDL_audio_resampler_filter.h */; };
 		F32DDAD32AB795A30041EAA5 /* SDL_audioqueue.h in Headers */ = {isa = PBXBuildFile; fileRef = F32DDACD2AB795A30041EAA5 /* SDL_audioqueue.h */; };
 		F32DDAD42AB795A30041EAA5 /* SDL_audioresample.c in Sources */ = {isa = PBXBuildFile; fileRef = F32DDACE2AB795A30041EAA5 /* SDL_audioresample.c */; };
 		F34B9895291DEFF500AAC96E /* SDL_hidapi_steam.c in Sources */ = {isa = PBXBuildFile; fileRef = A75FDAAC23E2795C00529352 /* SDL_hidapi_steam.c */; };
@@ -916,7 +915,6 @@
 		F32DDAC92AB795A30041EAA5 /* SDL_audio_channel_converters.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audio_channel_converters.h; sourceTree = "<group>"; };
 		F32DDACA2AB795A30041EAA5 /* SDL_audioresample.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audioresample.h; sourceTree = "<group>"; };
 		F32DDACB2AB795A30041EAA5 /* SDL_audioqueue.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audioqueue.c; sourceTree = "<group>"; };
-		F32DDACC2AB795A30041EAA5 /* SDL_audio_resampler_filter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audio_resampler_filter.h; sourceTree = "<group>"; };
 		F32DDACD2AB795A30041EAA5 /* SDL_audioqueue.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audioqueue.h; sourceTree = "<group>"; };
 		F32DDACE2AB795A30041EAA5 /* SDL_audioresample.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audioresample.c; sourceTree = "<group>"; };
 		F362B9152B3349E200D30B94 /* controller_list.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = controller_list.h; sourceTree = "<group>"; };
@@ -1982,7 +1980,6 @@
 				A7D8A87023E2513F00DCD162 /* dummy */,
 				A7D8A87323E2513F00DCD162 /* SDL_audio_c.h */,
 				F32DDAC92AB795A30041EAA5 /* SDL_audio_channel_converters.h */,
-				F32DDACC2AB795A30041EAA5 /* SDL_audio_resampler_filter.h */,
 				A7D8A8B823E2513F00DCD162 /* SDL_audio.c */,
 				A7D8A8A123E2513F00DCD162 /* SDL_audiocvt.c */,
 				A7D8A87723E2513F00DCD162 /* SDL_audiodev_c.h */,
@@ -2295,7 +2292,6 @@
 				F3F7D8ED2933074E00816151 /* SDL_audio.h in Headers */,
 				A7D8B7A023E2514400DCD162 /* SDL_audio_c.h in Headers */,
 				F32DDACF2AB795A30041EAA5 /* SDL_audio_channel_converters.h in Headers */,
-				F32DDAD22AB795A30041EAA5 /* SDL_audio_resampler_filter.h in Headers */,
 				A7D8B7B223E2514400DCD162 /* SDL_audiodev_c.h in Headers */,
 				F32DDAD32AB795A30041EAA5 /* SDL_audioqueue.h in Headers */,
 				F32DDAD02AB795A30041EAA5 /* SDL_audioresample.h in Headers */,

build-scripts/gen_audio_resampler_filter.c

@@ -1,149 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2024 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>
-
-#ifndef M_PI
-    #define M_PI 3.14159265358979323846
-#endif
-
-#define RESAMPLER_ZERO_CROSSINGS 5
-#define RESAMPLER_BITS_PER_SAMPLE 16
-#define RESAMPLER_BITS_PER_ZERO_CROSSING  ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)
-#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING  (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)
-#define RESAMPLER_FILTER_SIZE (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS)
-
-/* 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 = 1.0;
-    double f = 1.0;
-    int i = 1;
-
-    while (1) {
-        const double diff = pow(xdiv2, i * 2) / pow(f, 2);
-        if (diff < 1.0e-21) {
-            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(double *table, const int tablelen, const double beta)
-{
-    const double bessel_beta = bessel(beta);
-    int i;
-
-    table[0] = 1.0;
-
-    for (i = 1; i < tablelen; i++) {
-        const double kaiser = bessel(beta * sqrt(1.0 - pow((double)i / (double)(tablelen), 2.0))) / bessel_beta;
-        const double x = (((double) i) / ((double) RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) * M_PI;
-        table[i] = kaiser * (sin(x) / x);
-    }
-}
-
-static double ResamplerFilter[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, RESAMPLER_FILTER_SIZE, beta);
-}
-
-int main(void)
-{
-    int i, j;
-
-    PrepareResampleFilter();
-
-    printf(
-        "/*\n"
-        "  Simple DirectMedia Layer\n"
-        "  Copyright (C) 1997-2024 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_BITS_PER_ZERO_CROSSING ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)\n"
-        "#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)\n"
-        "#define RESAMPLER_FILTER_SIZE (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS)\n"
-        "\n", RESAMPLER_ZERO_CROSSINGS, RESAMPLER_BITS_PER_SAMPLE
-    );
-
-    printf("static const float ResamplerFilter[RESAMPLER_FILTER_SIZE] = {");
-    for (i = 0; i < RESAMPLER_FILTER_SIZE; i++) {
-        j = (i % RESAMPLER_ZERO_CROSSINGS) * RESAMPLER_SAMPLES_PER_ZERO_CROSSING + (i / RESAMPLER_ZERO_CROSSINGS);
-        printf("%s%12.9ff,", (i % RESAMPLER_ZERO_CROSSINGS) ? "" : "\n    ", ResamplerFilter[j]);
-    }
-    printf("\n};\n\n");
-
-    return 0;
-}

src/audio/SDL_audio_resampler_filter.h

@@ -1,544 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2024 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.
-*/
-
-// DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c
-
-#define RESAMPLER_ZERO_CROSSINGS 5
-#define RESAMPLER_BITS_PER_SAMPLE 16
-#define RESAMPLER_BITS_PER_ZERO_CROSSING ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)
-#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)
-#define RESAMPLER_FILTER_SIZE (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS)
-
-static const float ResamplerFilter[RESAMPLER_FILTER_SIZE] = {
-     1.000000000f, 0.000000000f,-0.000000000f, 0.000000000f,-0.000000000f,
-     0.999993165f,-0.001679888f, 0.000529080f,-0.000151513f, 0.000027455f,
-     0.999972661f,-0.003351212f, 0.001055794f,-0.000302183f, 0.000054683f,
-     0.999938488f,-0.005013955f, 0.001580128f,-0.000452009f, 0.000081685f,
-     0.999890647f,-0.006668099f, 0.002102071f,-0.000600987f, 0.000108459f,
-     0.999829139f,-0.008313629f, 0.002621611f,-0.000749115f, 0.000135007f,
-     0.999753966f,-0.009950528f, 0.003138734f,-0.000896389f, 0.000161328f,
-     0.999665131f,-0.011578779f, 0.003653429f,-0.001042807f, 0.000187423f,
-     0.999562634f,-0.013198368f, 0.004165684f,-0.001188367f, 0.000213291f,
-     0.999446480f,-0.014809279f, 0.004675488f,-0.001333066f, 0.000238933f,
-     0.999316672f,-0.016411497f, 0.005182828f,-0.001476901f, 0.000264348f,
-     0.999173212f,-0.018005007f, 0.005687694f,-0.001619871f, 0.000289537f,
-     0.999016105f,-0.019589795f, 0.006190074f,-0.001761971f, 0.000314501f,
-     0.998845356f,-0.021165846f, 0.006689957f,-0.001903201f, 0.000339239f,
-     0.998660968f,-0.022733147f, 0.007187332f,-0.002043558f, 0.000363751f,
-     0.998462946f,-0.024291684f, 0.007682189f,-0.002183039f, 0.000388037f,
-     0.998251297f,-0.025841443f, 0.008174516f,-0.002321643f, 0.000412099f,
-     0.998026026f,-0.027382413f, 0.008664303f,-0.002459367f, 0.000435935f,
-     0.997787138f,-0.028914579f, 0.009151540f,-0.002596209f, 0.000459547f,
-     0.997534641f,-0.030437930f, 0.009636217f,-0.002732167f, 0.000482934f,
-     0.997268542f,-0.031952453f, 0.010118324f,-0.002867240f, 0.000506097f,
-     0.996988847f,-0.033458137f, 0.010597850f,-0.003001425f, 0.000529036f,
-     0.996695563f,-0.034954970f, 0.011074786f,-0.003134721f, 0.000551752f,
-     0.996388700f,-0.036442941f, 0.011549123f,-0.003267125f, 0.000574244f,
-     0.996068266f,-0.037922039f, 0.012020851f,-0.003398637f, 0.000596512f,
-     0.995734268f,-0.039392253f, 0.012489961f,-0.003529253f, 0.000618558f,
-     0.995386717f,-0.040853574f, 0.012956443f,-0.003658973f, 0.000640382f,
-     0.995025621f,-0.042305990f, 0.013420290f,-0.003787796f, 0.000661984f,
-     0.994650990f,-0.043749493f, 0.013881491f,-0.003915718f, 0.000683363f,
-     0.994262835f,-0.045184072f, 0.014340039f,-0.004042740f, 0.000704522f,
-     0.993861166f,-0.046609719f, 0.014795924f,-0.004168860f, 0.000725459f,
-     0.993445994f,-0.048026424f, 0.015249139f,-0.004294075f, 0.000746176f,
-     0.993017331f,-0.049434180f, 0.015699676f,-0.004418386f, 0.000766672f,
-     0.992575187f,-0.050832978f, 0.016147525f,-0.004541790f, 0.000786949f,
-     0.992119574f,-0.052222809f, 0.016592680f,-0.004664287f, 0.000807006f,
-     0.991650506f,-0.053603666f, 0.017035133f,-0.004785875f, 0.000826844f,
-     0.991167995f,-0.054975543f, 0.017474875f,-0.004906553f, 0.000846464f,
-     0.990672054f,-0.056338431f, 0.017911900f,-0.005026320f, 0.000865865f,
-     0.990162696f,-0.057692323f, 0.018346201f,-0.005145175f, 0.000885049f,
-     0.989639935f,-0.059037214f, 0.018777770f,-0.005263117f, 0.000904016f,
-     0.989103786f,-0.060373097f, 0.019206599f,-0.005380146f, 0.000922766f,
-     0.988554262f,-0.061699966f, 0.019632684f,-0.005496260f, 0.000941300f,
-     0.987991380f,-0.063017815f, 0.020056015f,-0.005611458f, 0.000959619f,
-     0.987415153f,-0.064326639f, 0.020476588f,-0.005725741f, 0.000977722f,
-     0.986825598f,-0.065626433f, 0.020894396f,-0.005839106f, 0.000995611f,
-     0.986222730f,-0.066917192f, 0.021309432f,-0.005951554f, 0.001013285f,
-     0.985606567f,-0.068198912f, 0.021721690f,-0.006063084f, 0.001030746f,
-     0.984977124f,-0.069471588f, 0.022131165f,-0.006173695f, 0.001047994f,
-     0.984334418f,-0.070735217f, 0.022537850f,-0.006283387f, 0.001065030f,
-     0.983678468f,-0.071989794f, 0.022941741f,-0.006392159f, 0.001081853f,
-     0.983009290f,-0.073235317f, 0.023342830f,-0.006500011f, 0.001098466f,
-     0.982326903f,-0.074471782f, 0.023741114f,-0.006606943f, 0.001114868f,
-     0.981631326f,-0.075699186f, 0.024136587f,-0.006712954f, 0.001131059f,
-     0.980922577f,-0.076917527f, 0.024529243f,-0.006818044f, 0.001147042f,
-     0.980200675f,-0.078126804f, 0.024919078f,-0.006922213f, 0.001162815f,
-     0.979465640f,-0.079327013f, 0.025306087f,-0.007025460f, 0.001178380f,
-     0.978717491f,-0.080518153f, 0.025690266f,-0.007127786f, 0.001193738f,
-     0.977956250f,-0.081700223f, 0.026071609f,-0.007229191f, 0.001208889f,
-     0.977181936f,-0.082873221f, 0.026450113f,-0.007329674f, 0.001223833f,
-     0.976394570f,-0.084037148f, 0.026825773f,-0.007429235f, 0.001238572f,
-     0.975594175f,-0.085192002f, 0.027198586f,-0.007527875f, 0.001253106f,
-     0.974780770f,-0.086337783f, 0.027568547f,-0.007625593f, 0.001267436f,
-     0.973954379f,-0.087474491f, 0.027935652f,-0.007722391f, 0.001281562f,
-     0.973115024f,-0.088602126f, 0.028299898f,-0.007818267f, 0.001295485f,
-     0.972262727f,-0.089720690f, 0.028661282f,-0.007913223f, 0.001309207f,
-     0.971397512f,-0.090830182f, 0.029019799f,-0.008007258f, 0.001322726f,
-     0.970519401f,-0.091930604f, 0.029375448f,-0.008100373f, 0.001336045f,
-     0.969628418f,-0.093021958f, 0.029728224f,-0.008192568f, 0.001349164f,
-     0.968724588f,-0.094104245f, 0.030078125f,-0.008283845f, 0.001362084f,
-     0.967807935f,-0.095177467f, 0.030425148f,-0.008374202f, 0.001374806f,
-     0.966878483f,-0.096241627f, 0.030769290f,-0.008463642f, 0.001387329f,
-     0.965936258f,-0.097296726f, 0.031110550f,-0.008552163f, 0.001399656f,
-     0.964981285f,-0.098342768f, 0.031448923f,-0.008639768f, 0.001411786f,
-     0.964013590f,-0.099379756f, 0.031784409f,-0.008726456f, 0.001423721f,
-     0.963033199f,-0.100407693f, 0.032117005f,-0.008812229f, 0.001435461f,
-     0.962040138f,-0.101426582f, 0.032446709f,-0.008897086f, 0.001447008f,
-     0.961034434f,-0.102436428f, 0.032773519f,-0.008981030f, 0.001458361f,
-     0.960016114f,-0.103437235f, 0.033097434f,-0.009064060f, 0.001469522f,
-     0.958985206f,-0.104429007f, 0.033418451f,-0.009146178f, 0.001480492f,
-     0.957941737f,-0.105411749f, 0.033736571f,-0.009227385f, 0.001491271f,
-     0.956885736f,-0.106385466f, 0.034051790f,-0.009307680f, 0.001501860f,
-     0.955817231f,-0.107350163f, 0.034364109f,-0.009387067f, 0.001512261f,
-     0.954736250f,-0.108305845f, 0.034673526f,-0.009465545f, 0.001522473f,
-     0.953642823f,-0.109252518f, 0.034980040f,-0.009543115f, 0.001532497f,
-     0.952536979f,-0.110190189f, 0.035283651f,-0.009619779f, 0.001542336f,
-     0.951418748f,-0.111118864f, 0.035584357f,-0.009695538f, 0.001551988f,
-     0.950288160f,-0.112038548f, 0.035882158f,-0.009770393f, 0.001561456f,
-     0.949145245f,-0.112949250f, 0.036177055f,-0.009844346f, 0.001570741f,
-     0.947990034f,-0.113850976f, 0.036469046f,-0.009917397f, 0.001579842f,
-     0.946822559f,-0.114743733f, 0.036758132f,-0.009989548f, 0.001588761f,
-     0.945642850f,-0.115627529f, 0.037044312f,-0.010060800f, 0.001597498f,
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-     0.071979836f,-0.022327686f, 0.006528191f,-0.001246443f, 0.000062412f,
-     0.069982467f,-0.021717836f, 0.006346731f,-0.001210271f, 0.000060225f,
-     0.067992764f,-0.021109739f, 0.006165955f,-0.001174315f, 0.000058072f,
-     0.066010755f,-0.020503413f, 0.005985866f,-0.001138576f, 0.000055951f,
-     0.064036468f,-0.019898876f, 0.005806472f,-0.001103055f, 0.000053862f,
-     0.062069932f,-0.019296147f, 0.005627775f,-0.001067751f, 0.000051807f,
-     0.060111174f,-0.018695244f, 0.005449782f,-0.001032667f, 0.000049783f,
-     0.058160222f,-0.018096185f, 0.005272497f,-0.000997801f, 0.000047792f,
-     0.056217102f,-0.017498988f, 0.005095926f,-0.000963156f, 0.000045833f,
-     0.054281842f,-0.016903671f, 0.004920073f,-0.000928730f, 0.000043905f,
-     0.052354468f,-0.016310251f, 0.004744943f,-0.000894525f, 0.000042010f,
-     0.050435006f,-0.015718746f, 0.004570540f,-0.000860541f, 0.000040146f,
-     0.048523483f,-0.015129173f, 0.004396870f,-0.000826779f, 0.000038313f,
-     0.046619924f,-0.014541549f, 0.004223937f,-0.000793238f, 0.000036511f,
-     0.044724355f,-0.013955892f, 0.004051745f,-0.000759920f, 0.000034740f,
-     0.042836800f,-0.013372217f, 0.003880299f,-0.000726824f, 0.000033000f,
-     0.040957285f,-0.012790543f, 0.003709604f,-0.000693951f, 0.000031291f,
-     0.039085833f,-0.012210884f, 0.003539663f,-0.000661301f, 0.000029612f,
-     0.037222470f,-0.011633259f, 0.003370481f,-0.000628876f, 0.000027964f,
-     0.035367219f,-0.011057682f, 0.003202062f,-0.000596674f, 0.000026345f,
-     0.033520103f,-0.010484170f, 0.003034411f,-0.000564696f, 0.000024756f,
-     0.031681147f,-0.009912739f, 0.002867532f,-0.000532943f, 0.000023197f,
-     0.029850373f,-0.009343404f, 0.002701428f,-0.000501414f, 0.000021668f,
-     0.028027804f,-0.008776182f, 0.002536104f,-0.000470111f, 0.000020168f,
-     0.026213462f,-0.008211087f, 0.002371564f,-0.000439032f, 0.000018697f,
-     0.024407370f,-0.007648135f, 0.002207811f,-0.000408180f, 0.000017255f,
-     0.022609549f,-0.007087341f, 0.002044849f,-0.000377553f, 0.000015842f,
-     0.020820021f,-0.006528720f, 0.001882682f,-0.000347151f, 0.000014457f,
-     0.019038808f,-0.005972286f, 0.001721315f,-0.000316976f, 0.000013101f,
-     0.017265930f,-0.005418056f, 0.001560750f,-0.000287027f, 0.000011773f,
-     0.015501408f,-0.004866042f, 0.001400991f,-0.000257305f, 0.000010473f,
-     0.013745262f,-0.004316260f, 0.001242041f,-0.000227808f, 0.000009201f,
-     0.011997513f,-0.003768723f, 0.001083905f,-0.000198539f, 0.000007956f,
-     0.010258180f,-0.003223445f, 0.000926585f,-0.000169495f, 0.000006739f,
-     0.008527283f,-0.002680442f, 0.000770086f,-0.000140679f, 0.000005549f,
-     0.006804842f,-0.002139725f, 0.000614409f,-0.000112090f, 0.000004386f,
-     0.005090874f,-0.001601309f, 0.000459559f,-0.000083727f, 0.000003250f,
-     0.003385399f,-0.001065208f, 0.000305539f,-0.000055591f, 0.000002140f,
-     0.001688435f,-0.000531434f, 0.000152351f,-0.000027682f, 0.000001057f,
-};
-

src/audio/SDL_audioresample.c

@@ -21,208 +21,566 @@
 #include "SDL_internal.h"
 
 #include "SDL_sysaudio.h"
+
 #include "SDL_audioresample.h"
 
 // SDL's resampler uses a "bandlimited interpolation" algorithm:
 //     https://ccrma.stanford.edu/~jos/resample/
 
-#include "SDL_audio_resampler_filter.h"
+// TODO: Support changing this at runtime?
+#if defined(SDL_SSE_INTRINSICS) || defined(SDL_NEON_INTRINSICS)
+// In <current year>, SSE is basically mandatory anyway
+// We want RESAMPLER_SAMPLES_PER_FRAME to be a multiple of 4, to make SIMD easier
+#define RESAMPLER_ZERO_CROSSINGS 6
+#else
+#define RESAMPLER_ZERO_CROSSINGS 5
+#endif
+
+#define RESAMPLER_SAMPLES_PER_FRAME (RESAMPLER_ZERO_CROSSINGS * 2)
 
 // For a given srcpos, `srcpos + frame` are sampled, where `-RESAMPLER_ZERO_CROSSINGS < frame <= RESAMPLER_ZERO_CROSSINGS`.
 // Note, when upsampling, it is also possible to start sampling from `srcpos = -1`.
 #define RESAMPLER_MAX_PADDING_FRAMES (RESAMPLER_ZERO_CROSSINGS + 1)
 
-#define RESAMPLER_FILTER_INTERP_BITS  (32 - RESAMPLER_BITS_PER_ZERO_CROSSING)
-#define RESAMPLER_FILTER_INTERP_RANGE (1 << RESAMPLER_FILTER_INTERP_BITS)
+// More bits gives more precision, at the cost of a larger table.
+#define RESAMPLER_BITS_PER_ZERO_CROSSING    3
+#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)
+#define RESAMPLER_FILTER_INTERP_BITS        (32 - RESAMPLER_BITS_PER_ZERO_CROSSING)
+#define RESAMPLER_FILTER_INTERP_RANGE       (1 << RESAMPLER_FILTER_INTERP_BITS)
 
-#define RESAMPLER_SAMPLES_PER_FRAME (RESAMPLER_ZERO_CROSSINGS * 2)
+// ResampleFrame is just a vector/matrix/matrix multiplication.
+// It performs cubic interpolation of the filter, then multiplies that with the input.
+// dst = [1, frac, frac^2, frac^3] * filter * src
+
+// Cubic Polynomial
+typedef union Cubic
+{
+    float v[4];
+
+#ifdef SDL_SSE_INTRINSICS
+    // Aligned loads can be used directly as memory operands for mul/add
+    __m128 v128;
+#endif
+
+#ifdef SDL_NEON_INTRINSICS
+    float32x4_t v128;
+#endif
 
-#define RESAMPLER_FULL_FILTER_SIZE (RESAMPLER_SAMPLES_PER_FRAME * (RESAMPLER_SAMPLES_PER_ZERO_CROSSING + 1))
+} Cubic;
 
-static void ResampleFrame_Scalar(const float *src, float *dst, const float *raw_filter, float interp, int chans)
+static void ResampleFrame_Generic(const float *src, float *dst, const Cubic *filter, float frac, int chans)
 {
+    const float frac2 = frac * frac;
+    const float frac3 = frac * frac2;
+
     int i, chan;
+    float scales[RESAMPLER_SAMPLES_PER_FRAME];
+
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i, ++filter) {
+        scales[i] = filter->v[0] + (filter->v[1] * frac) + (filter->v[2] * frac2) + (filter->v[3] * frac3);
+    }
+
+    for (chan = 0; chan < chans; ++chan) {
+        float out = 0.0f;
+
+        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i) {
+            out += src[i * chans + chan] * scales[i];
+        }
+
+        dst[chan] = out;
+    }
+}
+
+static void ResampleFrame_Mono(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+    const float frac2 = frac * frac;
+    const float frac3 = frac * frac2;
+
+    int i;
+    float out = 0.0f;
+
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i, ++filter) {
+        // Interpolate between the nearest two filters
+        const float scale = filter->v[0] + (filter->v[1] * frac) + (filter->v[2] * frac2) + (filter->v[3] * frac3);
+
+        out += src[i] * scale;
+    }
+
+    dst[0] = out;
+}
+
+static void ResampleFrame_Stereo(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+    const float frac2 = frac * frac;
+    const float frac3 = frac * frac2;
+
+    int i;
+    float out0 = 0.0f;
+    float out1 = 0.0f;
 
-    float filter[RESAMPLER_SAMPLES_PER_FRAME];
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i, ++filter) {
+        // Interpolate between the nearest two filters
+        const float scale = filter->v[0] + (filter->v[1] * frac) + (filter->v[2] * frac2) + (filter->v[3] * frac3);
 
-    // Interpolate between the nearest two filters
-    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
-        filter[i] = (raw_filter[i] * (1.0f - interp)) + (raw_filter[i + RESAMPLER_SAMPLES_PER_FRAME] * interp);
+        out0 += src[i * 2 + 0] * scale;
+        out1 += src[i * 2 + 1] * scale;
+    }
+
+    dst[0] = out0;
+    dst[1] = out1;
+}
+
+#ifdef SDL_SSE_INTRINSICS
+#define sdl_madd_ps(a, b, c) _mm_add_ps(a, _mm_mul_ps(b, c)) // Not-so-fused multiply-add
+
+static void SDL_TARGETING("sse") ResampleFrame_Generic_SSE(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+#if RESAMPLER_SAMPLES_PER_FRAME != 12
+#error Invalid samples per frame
+#endif
+
+    __m128 f0, f1, f2;
+
+    {
+        const __m128 frac1 = _mm_set1_ps(frac);
+        const __m128 frac2 = _mm_mul_ps(frac1, frac1);
+        const __m128 frac3 = _mm_mul_ps(frac1, frac2);
+
+/* Transposed in SetupAudioResampler */
+#define X(out)                                                                                                                        \
+    out = sdl_madd_ps(sdl_madd_ps(sdl_madd_ps(filter[0].v128, filter[1].v128, frac1), filter[2].v128, frac2), filter[3].v128, frac3); \
+    filter += 4
+
+        X(f0);
+        X(f1);
+        X(f2);
+
+#undef X
     }
 
     if (chans == 2) {
-        float out[2];
-        out[0] = 0.0f;
-        out[1] = 0.0f;
-
-        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
-            const float scale = filter[i];
-            out[0] += src[i * 2 + 0] * scale;
-            out[1] += src[i * 2 + 1] * scale;
-        }
+        // Duplicate each of the filter elements and multiply by the input
+        // Use two accumulators to improve throughput
+        __m128 out0 = _mm_mul_ps(_mm_loadu_ps(src + 0), _mm_unpacklo_ps(f0, f0));
+        __m128 out1 = _mm_mul_ps(_mm_loadu_ps(src + 4), _mm_unpackhi_ps(f0, f0));
+        out0 = sdl_madd_ps(out0, _mm_loadu_ps(src + 8), _mm_unpacklo_ps(f1, f1));
+        out1 = sdl_madd_ps(out1, _mm_loadu_ps(src + 12), _mm_unpackhi_ps(f1, f1));
+        out0 = sdl_madd_ps(out0, _mm_loadu_ps(src + 16), _mm_unpacklo_ps(f2, f2));
+        out1 = sdl_madd_ps(out1, _mm_loadu_ps(src + 20), _mm_unpackhi_ps(f2, f2));
+
+        // Add the accumulators together
+        __m128 out = _mm_add_ps(out0, out1);
+
+        // Add the lower and upper pairs together
+        out = _mm_add_ps(out, _mm_movehl_ps(out, out));
 
-        dst[0] = out[0];
-        dst[1] = out[1];
+        // Store the result
+        _mm_storel_pi((__m64 *)dst, out);
         return;
     }
 
     if (chans == 1) {
-        float out = 0.0f;
+        // Multiply the filter by the input
+        __m128 out = _mm_mul_ps(f0, _mm_loadu_ps(src + 0));
+        out = sdl_madd_ps(out, f1, _mm_loadu_ps(src + 4));
+        out = sdl_madd_ps(out, f2, _mm_loadu_ps(src + 8));
 
-        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
-            out += src[i] * filter[i];
-        }
+        // Horizontal sum
+        __m128 shuf = _mm_shuffle_ps(out, out, _MM_SHUFFLE(2, 3, 0, 1));
+        out = _mm_add_ps(out, shuf);
+        out = _mm_add_ss(out, _mm_movehl_ps(shuf, out));
 
-        dst[0] = out;
+        _mm_store_ss(dst, out);
         return;
     }
 
-    for (chan = 0; chan < chans; chan++) {
-        float f = 0.0f;
+    int chan = 0;
 
-        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
-            f += src[i * chans + chan] * filter[i];
-        }
+    // Process 4 channels at once
+    for (; chan + 4 <= chans; chan += 4) {
+        const float *in = &src[chan];
+        __m128 out0 = _mm_setzero_ps();
+        __m128 out1 = _mm_setzero_ps();
+
+#define X(a, b, out)                                                                         \
+    out = sdl_madd_ps(out, _mm_loadu_ps(in), _mm_shuffle_ps(a, a, _MM_SHUFFLE(b, b, b, b))); \
+    in += chans
+
+#define Y(a)       \
+    X(a, 0, out0); \
+    X(a, 1, out1); \
+    X(a, 2, out0); \
+    X(a, 3, out1)
+
+        Y(f0);
+        Y(f1);
+        Y(f2);
 
-        dst[chan] = f;
+#undef X
+#undef Y
+
+        // Add the accumulators together
+        __m128 out = _mm_add_ps(out0, out1);
+
+        _mm_storeu_ps(&dst[chan], out);
+    }
+
+    // Process the remaining channels one at a time.
+    // Channel counts 1,2,4,8 are already handled above, leaving 3,5,6,7 to deal with (looping 3,1,2,3 times).
+    // Without vgatherdps (AVX2), this gets quite messy.
+    for (; chan < chans; ++chan) {
+        const float *in = &src[chan];
+        __m128 v0, v1, v2;
+
+#define X(x)                                                                         \
+    x = _mm_unpacklo_ps(_mm_load_ss(in), _mm_load_ss(in + chans));                   \
+    in += chans + chans;                                                             \
+    x = _mm_movelh_ps(x, _mm_unpacklo_ps(_mm_load_ss(in), _mm_load_ss(in + chans))); \
+    in += chans + chans
+
+        X(v0);
+        X(v1);
+        X(v2);
+
+#undef X
+
+        __m128 out = _mm_mul_ps(f0, v0);
+        out = sdl_madd_ps(out, f1, v1);
+        out = sdl_madd_ps(out, f2, v2);
+
+        // Horizontal sum
+        __m128 shuf = _mm_shuffle_ps(out, out, _MM_SHUFFLE(2, 3, 0, 1));
+        out = _mm_add_ps(out, shuf);
+        out = _mm_add_ss(out, _mm_movehl_ps(shuf, out));
+
+        _mm_store_ss(&dst[chan], out);
     }
 }
 
-#ifdef SDL_SSE_INTRINSICS
-static void SDL_TARGETING("sse") ResampleFrame_SSE(const float *src, float *dst, const float *raw_filter, float interp, int chans)
+#undef sdl_madd_ps
+#endif
+
+#ifdef SDL_NEON_INTRINSICS
+static void ResampleFrame_Generic_NEON(const float *src, float *dst, const Cubic *filter, float frac, int chans)
 {
-#if RESAMPLER_SAMPLES_PER_FRAME != 10
+#if RESAMPLER_SAMPLES_PER_FRAME != 12
 #error Invalid samples per frame
 #endif
 
-    // Load the filter
-    __m128 f0 = _mm_loadu_ps(raw_filter + 0);
-    __m128 f1 = _mm_loadu_ps(raw_filter + 4);
-    __m128 f2 = _mm_loadl_pi(_mm_setzero_ps(), (const __m64 *)(raw_filter + 8));
+    float32x4_t f0, f1, f2;
 
-    __m128 g0 = _mm_loadu_ps(raw_filter + 10);
-    __m128 g1 = _mm_loadu_ps(raw_filter + 14);
-    __m128 g2 = _mm_loadl_pi(_mm_setzero_ps(), (const __m64 *)(raw_filter + 18));
+    {
+        const float32x4_t frac1 = vdupq_n_f32(frac);
+        const float32x4_t frac2 = vmulq_f32(frac1, frac1);
+        const float32x4_t frac3 = vmulq_f32(frac1, frac2);
 
-    __m128 interp1 = _mm_set1_ps(interp);
-    __m128 interp2 = _mm_sub_ps(_mm_set1_ps(1.0f), _mm_set1_ps(interp));
+/* Transposed in SetupAudioResampler */
+#define X(out)                                                                                                                  \
+    out = vmlaq_f32(vmlaq_f32(vmlaq_f32(filter[0].v128, filter[1].v128, frac1), filter[2].v128, frac2), filter[3].v128, frac3); \
+    filter += 4
 
-    // Linear interpolate the filter
-    f0 = _mm_add_ps(_mm_mul_ps(f0, interp2), _mm_mul_ps(g0, interp1));
-    f1 = _mm_add_ps(_mm_mul_ps(f1, interp2), _mm_mul_ps(g1, interp1));
-    f2 = _mm_add_ps(_mm_mul_ps(f2, interp2), _mm_mul_ps(g2, interp1));
+        X(f0);
+        X(f1);
+        X(f2);
+
+#undef X
+    }
 
     if (chans == 2) {
-        // Duplicate each of the filter elements
-        g0 = _mm_unpackhi_ps(f0, f0);
-        f0 = _mm_unpacklo_ps(f0, f0);
-        g1 = _mm_unpackhi_ps(f1, f1);
-        f1 = _mm_unpacklo_ps(f1, f1);
-        f2 = _mm_unpacklo_ps(f2, f2);
+        float32x4x2_t g0 = vzipq_f32(f0, f0);
+        float32x4x2_t g1 = vzipq_f32(f1, f1);
+        float32x4x2_t g2 = vzipq_f32(f2, f2);
 
-        // Multiply the filter by the input
-        f0 = _mm_mul_ps(f0, _mm_loadu_ps(src + 0));
-        g0 = _mm_mul_ps(g0, _mm_loadu_ps(src + 4));
-        f1 = _mm_mul_ps(f1, _mm_loadu_ps(src + 8));
-        g1 = _mm_mul_ps(g1, _mm_loadu_ps(src + 12));
-        f2 = _mm_mul_ps(f2, _mm_loadu_ps(src + 16));
+        // Duplicate each of the filter elements and multiply by the input
+        // Use two accumulators to improve throughput
+        float32x4_t out0 = vmulq_f32(vld1q_f32(src + 0), g0.val[0]);
+        float32x4_t out1 = vmulq_f32(vld1q_f32(src + 4), g0.val[1]);
+        out0 = vmlaq_f32(out0, vld1q_f32(src + 8), g1.val[0]);
+        out1 = vmlaq_f32(out1, vld1q_f32(src + 12), g1.val[1]);
+        out0 = vmlaq_f32(out0, vld1q_f32(src + 16), g2.val[0]);
+        out1 = vmlaq_f32(out1, vld1q_f32(src + 20), g2.val[1]);
+
+        // Add the accumulators together
+        out0 = vaddq_f32(out0, out1);
 
-        // Calculate the sum
-        f0 = _mm_add_ps(_mm_add_ps(_mm_add_ps(f0, g0), _mm_add_ps(f1, g1)), f2);
-        f0 = _mm_add_ps(f0, _mm_movehl_ps(f0, f0));
+        // Add the lower and upper pairs together
+        float32x2_t out = vadd_f32(vget_low_f32(out0), vget_high_f32(out0));
 
         // Store the result
-        _mm_storel_pi((__m64 *)dst, f0);
+        vst1_f32(dst, out);
         return;
     }
 
     if (chans == 1) {
         // Multiply the filter by the input
-        f0 = _mm_mul_ps(f0, _mm_loadu_ps(src + 0));
-        f1 = _mm_mul_ps(f1, _mm_loadu_ps(src + 4));
-        f2 = _mm_mul_ps(f2, _mm_loadl_pi(_mm_setzero_ps(), (const __m64 *)(src + 8)));
+        float32x4_t out = vmulq_f32(f0, vld1q_f32(src + 0));
+        out = vmlaq_f32(out, f1, vld1q_f32(src + 4));
+        out = vmlaq_f32(out, f2, vld1q_f32(src + 8));
 
-        // Calculate the sum
-        f0 = _mm_add_ps(f0, f1);
-        f0 = _mm_add_ps(_mm_add_ps(f0, f2), _mm_movehl_ps(f0, f0));
-        f0 = _mm_add_ss(f0, _mm_shuffle_ps(f0, f0, _MM_SHUFFLE(1, 1, 1, 1)));
+        // Horizontal sum
+        float32x2_t sum = vadd_f32(vget_low_f32(out), vget_high_f32(out));
+        sum = vpadd_f32(sum, sum);
 
-        // Store the result
-        _mm_store_ss(dst, f0);
+        vst1_lane_f32(dst, sum, 0);
         return;
     }
 
-    float filter[RESAMPLER_SAMPLES_PER_FRAME];
-    _mm_storeu_ps(filter + 0, f0);
-    _mm_storeu_ps(filter + 4, f1);
-    _mm_storel_pi((__m64 *)(filter + 8), f2);
-
-    int i, chan = 0;
+    int chan = 0;
 
+    // Process 4 channels at once
     for (; chan + 4 <= chans; chan += 4) {
-        f0 = _mm_setzero_ps();
+        const float *in = &src[chan];
+        float32x4_t out0 = vdupq_n_f32(0);
+        float32x4_t out1 = vdupq_n_f32(0);
 
-        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
-            f0 = _mm_add_ps(f0, _mm_mul_ps(_mm_loadu_ps(&src[i * chans + chan]), _mm_load1_ps(&filter[i])));
-        }
+#define X(a, b, out)                                           \
+    out = vmlaq_f32(out, vld1q_f32(in), vdupq_lane_f32(a, b)); \
+    in += chans
 
-        _mm_storeu_ps(&dst[chan], f0);
-    }
+#define Y(a)                      \
+    X(vget_low_f32(a), 0, out0);  \
+    X(vget_low_f32(a), 1, out1);  \
+    X(vget_high_f32(a), 0, out0); \
+    X(vget_high_f32(a), 1, out1)
 
-    for (; chan < chans; chan++) {
-        f0 = _mm_setzero_ps();
+        Y(f0);
+        Y(f1);
+        Y(f2);
 
-        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
-            f0 = _mm_add_ss(f0, _mm_mul_ss(_mm_load_ss(&src[i * chans + chan]), _mm_load_ss(&filter[i])));
-        }
+#undef X
+#undef Y
+
+        // Add the accumulators together
+        float32x4_t out = vaddq_f32(out0, out1);
+
+        vst1q_f32(&dst[chan], out);
+    }
 
-        _mm_store_ss(&dst[chan], f0);
+    // Process the remaining channels one at a time.
+    // Channel counts 1,2,4,8 are already handled above, leaving 3,5,6,7 to deal with (looping 3,1,2,3 times).
+    for (; chan < chans; ++chan) {
+        const float *in = &src[chan];
+        float32x4_t v0, v1, v2;
+
+#define X(x)                      \
+    x = vld1q_dup_f32(in);        \
+    in += chans;                  \
+    x = vld1q_lane_f32(in, x, 1); \
+    in += chans;                  \
+    x = vld1q_lane_f32(in, x, 2); \
+    in += chans;                  \
+    x = vld1q_lane_f32(in, x, 3); \
+    in += chans
+
+        X(v0);
+        X(v1);
+        X(v2);
+
+#undef X
+
+        float32x4_t out = vmulq_f32(f0, v0);
+        out = vmlaq_f32(out, f1, v1);
+        out = vmlaq_f32(out, f2, v2);
+
+        // Horizontal sum
+        float32x2_t sum = vadd_f32(vget_low_f32(out), vget_high_f32(out));
+        sum = vpadd_f32(sum, sum);
+
+        vst1_lane_f32(&dst[chan], sum, 0);
     }
 }
 #endif
 
-static void (*ResampleFrame)(const float *src, float *dst, const float *raw_filter, float interp, int chans);
+// Calculate the cubic equation which passes through all four points.
+// https://en.wikipedia.org/wiki/Ordinary_least_squares
+// https://en.wikipedia.org/wiki/Polynomial_regression
+static void CubicLeastSquares(Cubic *coeffs, float y0, float y1, float y2, float y3)
+{
+    // Least squares matrix for xs = [0, 1/3, 2/3, 1]
+    // [  1.0   0.0   0.0  0.0 ]
+    // [ -5.5   9.0  -4.5  1.0 ]
+    // [  9.0 -22.5  18.0 -4.5 ]
+    // [ -4.5  13.5 -13.5  4.5 ]
+
+    coeffs->v[0] = y0;
+    coeffs->v[1] = -5.5f * y0 + 9.0f * y1 - 4.5f * y2 + y3;
+    coeffs->v[2] = 9.0f * y0 - 22.5f * y1 + 18.0f * y2 - 4.5f * y3;
+    coeffs->v[3] = -4.5f * y0 + 13.5f * y1 - 13.5f * y2 + 4.5f * y3;
+}
 
-static float FullResamplerFilter[RESAMPLER_FULL_FILTER_SIZE];
+// Zeroth-order modified Bessel function of the first kind
+// https://mathworld.wolfram.com/ModifiedBesselFunctionoftheFirstKind.html
+static float BesselI0(float x)
+{
+    float sum = 0.0f;
+    float i = 1.0f;
+    float t = 1.0f;
+    x *= x * 0.25f;
+
+    while (t >= sum * SDL_FLT_EPSILON) {
+        sum += t;
+        t *= x / (i * i);
+        ++i;
+    }
 
-void SDL_SetupAudioResampler(void)
+    return sum;
+}
+
+// Pre-calculate 180 degrees of sin(pi * x) / pi
+// The speedup from this isn't huge, but it also avoids precision issues.
+// If sinf isn't available, SDL_sinf just calls SDL_sin.
+// Know what SDL_sin(SDL_PI_F) equals? Not quite zero.
+static void SincTable(float *table, int len)
 {
-    static SDL_bool setup = SDL_FALSE;
-    if (setup) {
-        return;
+    int i;
+
+    for (i = 0; i < len; ++i) {
+        table[i] = SDL_sinf(i * (SDL_PI_F / len)) / SDL_PI_F;
     }
+}
+
+// Calculate Sinc(x/y), using a lookup table
+static float Sinc(float *table, int x, int y)
+{
+    float s = table[x % y];
+    s = ((x / y) & 1) ? -s : s;
+    return (s * y) / x;
+}
+
+Cubic ResamplerFilter[RESAMPLER_SAMPLES_PER_ZERO_CROSSING][RESAMPLER_SAMPLES_PER_FRAME];
+
+static void GenerateResamplerFilter()
+{
+    enum
+    {
+        // Generate samples at 3x the target resolution, so that we have samples at [0, 1/3, 2/3, 1] of each position
+        TABLE_SAMPLES_PER_ZERO_CROSSING = RESAMPLER_SAMPLES_PER_ZERO_CROSSING * 3,
+        TABLE_SIZE = RESAMPLER_ZERO_CROSSINGS * TABLE_SAMPLES_PER_ZERO_CROSSING,
+    };
+
+    // if dB > 50, beta=(0.1102 * (dB - 8.7)), according to Matlab.
+    const float dB = 80.0f;
+    const float beta = 0.1102f * (dB - 8.7f);
+    const float bessel_beta = BesselI0(beta);
+    const float lensqr = TABLE_SIZE * TABLE_SIZE;
 
-    // Build a table combining the left and right wings, for faster access
     int i, j;
 
+    float sinc[TABLE_SAMPLES_PER_ZERO_CROSSING];
+    SincTable(sinc, TABLE_SAMPLES_PER_ZERO_CROSSING);
+
+    // Generate one wing of the filter
+    // https://en.wikipedia.org/wiki/Kaiser_window
+    // https://en.wikipedia.org/wiki/Whittaker%E2%80%93Shannon_interpolation_formula
+    float filter[TABLE_SIZE + 1];
+    filter[0] = 1.0f;
+
+    for (i = 1; i <= TABLE_SIZE; ++i) {
+        float b = BesselI0(beta * SDL_sqrtf((lensqr - (i * i)) / lensqr)) / bessel_beta;
+        float s = Sinc(sinc, i, TABLE_SAMPLES_PER_ZERO_CROSSING);
+        filter[i] = b * s;
+    }
+
+    // Generate the coefficients for each point
+    // When interpolating, the fraction represents how far we are between input samples,
+    // so we need to align the filter by "moving" it to the right.
+    //
+    // For the left wing, this means interpolating "forwards" (away from the center)
+    // For the right wing, this means interpolating "backwards" (towards the center)
+    //
+    // The center of the filter is at the end of the left wing (RESAMPLER_ZERO_CROSSINGS - 1)
+    // The left wing is the filter, but reversed
+    // The right wing is the filter, but offset by 1
+    //
+    // Since the right wing is offset by 1, this just means we interpolate backwards
+    // between the same points, instead of forwards
+    // interp(p[n], p[n+1], t) = interp(p[n+1], p[n+1-1], 1 - t) = interp(p[n+1], p[n], 1 - t)
     for (i = 0; i < RESAMPLER_SAMPLES_PER_ZERO_CROSSING; ++i) {
-        for (j = 0; j < RESAMPLER_ZERO_CROSSINGS; j++) {
-            int lwing = (i * RESAMPLER_SAMPLES_PER_FRAME) + (RESAMPLER_ZERO_CROSSINGS - 1) - j;
-            int rwing = (RESAMPLER_FULL_FILTER_SIZE - 1) - lwing;
+        for (j = 0; j < RESAMPLER_ZERO_CROSSINGS; ++j) {
+            const float *ys = &filter[((j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING) + i) * 3];
+
+            Cubic *fwd = &ResamplerFilter[i][RESAMPLER_ZERO_CROSSINGS - j - 1];
+            Cubic *rev = &ResamplerFilter[RESAMPLER_SAMPLES_PER_ZERO_CROSSING - i - 1][RESAMPLER_ZERO_CROSSINGS + j];
 
-            float value = ResamplerFilter[(i * RESAMPLER_ZERO_CROSSINGS) + j];
-            FullResamplerFilter[lwing] = value;
-            FullResamplerFilter[rwing] = value;
+            // Calculate the cubic equation of the 4 points
+            CubicLeastSquares(fwd, ys[0], ys[1], ys[2], ys[3]);
+            CubicLeastSquares(rev, ys[3], ys[2], ys[1], ys[0]);
         }
     }
+}
+
+typedef void (*ResampleFrameFunc)(const float *src, float *dst, const Cubic *filter, float frac, int chans);
+static ResampleFrameFunc ResampleFrame[8];
 
-    for (i = 0; i < RESAMPLER_ZERO_CROSSINGS; ++i) {
-        int rwing = i + RESAMPLER_ZERO_CROSSINGS;
-        int lwing = (RESAMPLER_FULL_FILTER_SIZE - 1) - rwing;
+// Transpose 4x4 floats
+static void Transpose4x4(Cubic *data)
+{
+    int i, j;
 
-        FullResamplerFilter[lwing] = 0.0f;
-        FullResamplerFilter[rwing] = 0.0f;
+    Cubic temp[4] = { data[0], data[1], data[2], data[3] };
+
+    for (i = 0; i < 4; ++i) {
+        for (j = 0; j < 4; ++j) {
+            data[i].v[j] = temp[j].v[i];
+        }
     }
+}
+
+static void SetupAudioResampler(void)
+{
+    int i, j;
+    SDL_bool transpose = SDL_FALSE;
 
-    ResampleFrame = ResampleFrame_Scalar;
+    GenerateResamplerFilter();
 
 #ifdef SDL_SSE_INTRINSICS
     if (SDL_HasSSE()) {
-        ResampleFrame = ResampleFrame_SSE;
-    }
+        for (i = 0; i < 8; ++i) {
+            ResampleFrame[i] = ResampleFrame_Generic_SSE;
+        }
+        transpose = SDL_TRUE;
+    } else
 #endif
+#ifdef SDL_NEON_INTRINSICS
+    if (SDL_HasNEON()) {
+        for (i = 0; i < 8; ++i) {
+            ResampleFrame[i] = ResampleFrame_Generic_NEON;
+        }
+        transpose = SDL_TRUE;
+    } else
+#endif
+    {
+        for (i = 0; i < 8; ++i) {
+            ResampleFrame[i] = ResampleFrame_Generic;
+        }
+
+        ResampleFrame[0] = ResampleFrame_Mono;
+        ResampleFrame[1] = ResampleFrame_Stereo;
+    }
 
-    setup = SDL_TRUE;
+    if (transpose) {
+        // Transpose each set of 4 coefficients, to reduce work when resampling
+        for (i = 0; i < RESAMPLER_SAMPLES_PER_ZERO_CROSSING; ++i) {
+            for (j = 0; j + 4 <= RESAMPLER_SAMPLES_PER_FRAME; j += 4) {
+                Transpose4x4(&ResamplerFilter[i][j]);
+            }
+        }
+    }
+}
+
+void SDL_SetupAudioResampler(void)
+{
+    static SDL_SpinLock running = 0;
+
+    if (!ResampleFrame[0]) {
+        SDL_LockSpinlock(&running);
+
+        if (!ResampleFrame[0]) {
+            SetupAudioResampler();
+        }
+
+        SDL_UnlockSpinlock(&running);
+    }
 }
 
 Sint64 SDL_GetResampleRate(int src_rate, int dst_rate)
@@ -312,21 +670,24 @@ void SDL_ResampleAudio(int chans, const float *src, int inframes, float *dst, in
 {
     int i;
     Sint64 srcpos = *inout_resample_offset;
+    ResampleFrameFunc resample_frame = ResampleFrame[chans - 1];
 
     SDL_assert(resample_rate > 0);
 
-    for (i = 0; i < outframes; i++) {
+    src -= (RESAMPLER_ZERO_CROSSINGS - 1) * chans;
+
+    for (i = 0; i < outframes; ++i) {
         int srcindex = (int)(Sint32)(srcpos >> 32);
         Uint32 srcfraction = (Uint32)(srcpos & 0xFFFFFFFF);
         srcpos += resample_rate;
 
         SDL_assert(srcindex >= -1 && srcindex < inframes);
 
-        const float *filter = &FullResamplerFilter[(srcfraction >> RESAMPLER_FILTER_INTERP_BITS) * RESAMPLER_SAMPLES_PER_FRAME];
-        const float interp = (float)(srcfraction & (RESAMPLER_FILTER_INTERP_RANGE - 1)) * (1.0f / RESAMPLER_FILTER_INTERP_RANGE);
+        const Cubic *filter = ResamplerFilter[srcfraction >> RESAMPLER_FILTER_INTERP_BITS];
+        const float frac = (float)(srcfraction & (RESAMPLER_FILTER_INTERP_RANGE - 1)) * (1.0f / RESAMPLER_FILTER_INTERP_RANGE);
 
-        const float *frame = &src[(srcindex - (RESAMPLER_ZERO_CROSSINGS - 1)) * chans];
-        ResampleFrame(frame, dst, filter, interp, chans);
+        const float *frame = &src[srcindex * chans];
+        resample_frame(frame, dst, filter, frac, chans);
 
         dst += chans;
     }

test/testautomation_audio.c

@@ -844,7 +844,7 @@ static int audio_resampleLoss(void *arg)
     double signal_to_noise;
     double max_error;
   } test_specs[] = {
-    { 50, 440, 0, 44100, 48000, 80, 0.0009 },
+    { 50, 440, 0, 44100, 48000, 80, 0.0010 },
     { 50, 5000, SDL_PI_D / 2, 20000, 10000, 999, 0.0001 },
     { 50, 440, 0, 22050, 96000, 79, 0.0120 },
     { 50, 440, 0, 96000, 22050, 80, 0.0002 },
@@ -888,7 +888,7 @@ static int audio_resampleLoss(void *arg)
     tmpspec2.channels = num_channels;
     tmpspec2.freq = spec->rate_out;
     stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2);
-    SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out);
+    SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, %i, %i, SDL_AUDIO_F32, %i, %i)", num_channels, spec->rate_in, num_channels, spec->rate_out);
     SDLTest_AssertCheck(stream != NULL, "Expected SDL_CreateAudioStream to succeed.");
     if (stream == NULL) {
       return TEST_ABORTED;