fix(mix): reduce waveform smoothing to 15 ms; turn off DEBUG flags
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@@ -3,19 +3,20 @@ namespace DeepDrftContent.Processors;
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/// <summary>
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/// Loudness via root-mean-square amplitude per time bucket. Decodes signed PCM (8-bit unsigned,
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/// 16/24/32-bit signed little-endian), averages channels to mono, partitions the frames into
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/// equal time slices, takes the RMS of each slice, applies a ~50 ms envelope-follower smoothing
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/// equal time slices, takes the RMS of each slice, applies a ~15 ms envelope-follower smoothing
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/// so the contour reads as a smooth curve rather than a spikey polygon, then peak-normalizes so
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/// the loudest bucket is 1. No external audio dependency — operates directly on the WAV data-chunk bytes.
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/// </summary>
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public class RmsLoudnessAlgorithm : ILoudnessAlgorithm
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{
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/// <summary>
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/// Envelope-follower time constant, seconds. ~50 ms is the spec's smoothing target (Phase 10
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/// tuning): long enough to round off the per-bucket RMS spikes into a smooth ribbon contour,
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/// short enough that real loudness transients (kicks, drops) still read. Applied as a symmetric
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/// (forward+backward) one-pole filter so the smoothing introduces no time lag.
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/// Envelope-follower time constant, seconds. ~15 ms is the smoothing target (Phase 10
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/// tuning, reduced from 50 ms which was over-smoothed): long enough to round off the
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/// per-bucket RMS spikes into a smooth ribbon contour, short enough that real loudness
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/// transients (kicks, drops) still read. Applied as a symmetric (forward+backward) one-pole
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/// filter so the smoothing introduces no time lag.
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/// </summary>
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public const double SmoothingTimeConstantSeconds = 0.05;
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public const double SmoothingTimeConstantSeconds = 0.015;
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public double[] Compute(ReadOnlySpan<byte> pcmData, int channels, int sampleRate, int bitsPerSample, int bucketCount)
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{
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@@ -81,7 +82,7 @@ public class RmsLoudnessAlgorithm : ILoudnessAlgorithm
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}
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}
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// Envelope smoothing (~50 ms): round the spikey per-bucket RMS into a smooth contour before
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// Envelope smoothing (~15 ms): round the spikey per-bucket RMS into a smooth contour before
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// peak-normalization, so the rendered ribbon reads as a continuous curve, not faceted polygons.
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// Each bucket spans (totalSeconds / bucketCount) of audio; the filter coefficient is derived
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// from that against the time constant so the smoothing is duration-aware, not a fixed window.
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@@ -117,7 +118,7 @@ public class RmsLoudnessAlgorithm : ILoudnessAlgorithm
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/// Symmetric one-pole envelope smoothing over the per-bucket loudness, in place. A forward pass
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/// then a backward pass cancels the single-pole phase lag, so the smoothed contour stays aligned
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/// with the audio (no rightward time shift). The coefficient <c>a = exp(−bucketSeconds / τ)</c>
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/// gives a ~<paramref name="bucketSeconds"/>-relative response targeting the ~50 ms time constant:
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/// gives a ~<paramref name="bucketSeconds"/>-relative response targeting the ~15 ms time constant:
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/// each bucket blends <c>(1 − a)</c> of itself with <c>a</c> of the running envelope. A near-zero
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/// or non-finite bucket duration leaves the data untouched (nothing to smooth meaningfully).
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/// </summary>
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