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