Changed softfloat call in audio
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1 changed files with 1 additions and 3 deletions
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@ -1078,8 +1078,6 @@ static int sound_prefs_changed(void)
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return 0;
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}
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double softfloat_tan(double v);
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/* This computes the 1st order low-pass filter term b0.
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* The a1 term is 1.0 - b0. The center frequency marks the -3 dB point. */
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#ifndef M_PI
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@ -1095,7 +1093,7 @@ static float rc_calculate_a0(int sample_rate, int cutoff_freq)
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/* Compensate for the bilinear transformation. This allows us to specify the
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* stop frequency more exactly, but the filter becomes less steep further
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* from stopband. */
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omega = softfloat_tan(omega / 2.0) * 2.0;
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omega = tan(omega / 2.0) * 2.0;
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const float out = 1.0 / (1.0 + 1.0 / omega);
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return out;
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}
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