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author | RĂ©mi Verschelde <remi@verschelde.fr> | 2021-08-06 11:27:13 +0200 |
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committer | GitHub <noreply@github.com> | 2021-08-06 11:27:13 +0200 |
commit | d4e7a1af4416b6485ace5de19660d41cbe0fdb11 (patch) | |
tree | 716f241e54d1b60aeee74e01b8743f0a6e5484ac | |
parent | 18fb2afcacb401ebfa866b94f2be85da6f8feb0a (diff) | |
parent | 2d450c6f61e209c921eea352ed37afd98d11b034 (diff) |
Merge pull request #51082 from ellenhp/fix_cubic_resampling
-rw-r--r-- | servers/audio/audio_stream.cpp | 18 |
1 files changed, 9 insertions, 9 deletions
diff --git a/servers/audio/audio_stream.cpp b/servers/audio/audio_stream.cpp index aec6932326..aefea6723e 100644 --- a/servers/audio/audio_stream.cpp +++ b/servers/audio/audio_stream.cpp @@ -54,21 +54,21 @@ void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, for (int i = 0; i < p_frames; i++) { uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS); - // 4 point, 4th order optimal resampling algorithm from: http://yehar.com/blog/wp-content/uploads/2009/08/deip.pdf + //standard cubic interpolation (great quality/performance ratio) + //this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory. float mu = (mix_offset & FP_MASK) / float(FP_LEN); AudioFrame y0 = internal_buffer[idx - 3]; AudioFrame y1 = internal_buffer[idx - 2]; AudioFrame y2 = internal_buffer[idx - 1]; AudioFrame y3 = internal_buffer[idx - 0]; - AudioFrame even1 = y2 + y1, odd1 = y2 - y1; - AudioFrame even2 = y3 + y0, odd2 = y3 - y0; - AudioFrame c0 = even1 * 0.46835497211269561 + even2 * 0.03164502784253309; - AudioFrame c1 = odd1 * 0.56001293337091440 + odd2 * 0.14666238593949288; - AudioFrame c2 = even1 * -0.250038759826233691 + even2 * 0.25003876124297131; - AudioFrame c3 = odd1 * -0.49949850957839148 + odd2 * 0.16649935475113800; - AudioFrame c4 = even1 * 0.00016095224137360 + even2 * -0.00016095810460478; - p_buffer[i] = (((c4 * mu + c3) * mu + c2) * mu + c1) * mu + c0; + float mu2 = mu * mu; + AudioFrame a0 = 3 * y1 - 3 * y2 + y3 - y0; + AudioFrame a1 = 2 * y0 - 5 * y1 + 4 * y2 - y3; + AudioFrame a2 = y2 - y0; + AudioFrame a3 = 2 * y1; + + p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3) / 2; mix_offset += mix_increment; |