Revert "Implement a new resampling algorithm in AudioStreamPlaybackResampled"
This reverts commit b2264cb48b
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1 changed files with 9 additions and 9 deletions
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@ -54,21 +54,21 @@ void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale,
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for (int i = 0; i < p_frames; i++) {
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uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS);
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// 4 point, 4th order optimal resampling algorithm from: http://yehar.com/blog/wp-content/uploads/2009/08/deip.pdf
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//standard cubic interpolation (great quality/performance ratio)
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//this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory.
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float mu = (mix_offset & FP_MASK) / float(FP_LEN);
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AudioFrame y0 = internal_buffer[idx - 3];
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AudioFrame y1 = internal_buffer[idx - 2];
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AudioFrame y2 = internal_buffer[idx - 1];
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AudioFrame y3 = internal_buffer[idx - 0];
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AudioFrame even1 = y2 + y1, odd1 = y2 - y1;
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AudioFrame even2 = y3 + y0, odd2 = y3 - y0;
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AudioFrame c0 = even1 * 0.46835497211269561 + even2 * 0.03164502784253309;
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AudioFrame c1 = odd1 * 0.56001293337091440 + odd2 * 0.14666238593949288;
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AudioFrame c2 = even1 * -0.250038759826233691 + even2 * 0.25003876124297131;
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AudioFrame c3 = odd1 * -0.49949850957839148 + odd2 * 0.16649935475113800;
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AudioFrame c4 = even1 * 0.00016095224137360 + even2 * -0.00016095810460478;
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p_buffer[i] = (((c4 * mu + c3) * mu + c2) * mu + c1) * mu + c0;
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float mu2 = mu * mu;
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AudioFrame a0 = y3 - y2 - y0 + y1;
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AudioFrame a1 = y0 - y1 - a0;
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AudioFrame a2 = y2 - y0;
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AudioFrame a3 = y1;
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p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3);
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mix_offset += mix_increment;
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