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Remove chunking #4

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Remove chunking #4

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44 changes: 4 additions & 40 deletions src/dsp/mod.rs
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Original file line number Diff line number Diff line change
Expand Up @@ -42,53 +42,17 @@ impl PluginDsp {
}
}

// To take advantage of SIMD auto-vectorization, and for consistent parameter smoothing,
// audio is processed in "chunks" of 16 samples at a time. The number of samples requested
// by a host will generally be a multiple of 16, although the buffer may be truncated early
// in some cases.
//
// This approach is overly complex for such a simple use-case, but can be particularly
// useful for reducing unnecessary re-computation with many parameters.
let num_samples = buffer.samples();
let num_chunks = num_samples / 16;
let extra_samples = num_samples % 16;
let num_channels = buffer.input_count();

let (inputs, mut outputs) = buffer.split();
for chunk_start in (0..num_chunks).map(|i| i * 16) {
for sample_idx in 0..num_samples {
self.amplitude_range.process();

// Prepare the chunk's base amplitude value by placing it into a 16-element array, then
// linearly interpolate them towards the next value if the amplitude has recently been
// changed.
let mut chunk_amplitudes = [self.amplitude; 16];
if let Some(amplitude_range) = self.amplitude_range.get_new_value() {
let new_amplitude = amplitude_range * 2.;
let per_sample_difference = (new_amplitude - self.amplitude) / 16.;
chunk_amplitudes
.iter_mut()
.zip((0..16).map(|i| i as f32 * per_sample_difference))
.for_each(|(amplitude, difference)| *amplitude += difference);
if let Some(new_amplitude) = self.amplitude_range.get_new_value() {
self.amplitude = new_amplitude;
}

// Then, calculate each output sample by multiplying each input sample by its
// corresponding amplitude value.
for channel in 0..num_channels {
for (i, amplitude) in chunk_amplitudes.iter().enumerate() {
outputs[channel][chunk_start + i] =
inputs[channel][chunk_start + i] * amplitude;
}
}
}

// Finally, process the final <16 samples, if any.
for i in 0..extra_samples {
for channel in 0..num_channels {
// We could precompute extra interpolated amplitude values into a rollover buffer,
// but it's simpler to approximate by just reusing the last known amplitude value.
outputs[channel][num_chunks * 16 + i] =
inputs[channel][num_chunks * 16 + i] * self.amplitude;
for channel_idx in 0..num_channels {
outputs[channel_idx][sample_idx] = inputs[channel_idx][sample_idx] * self.amplitude;
}
}
}
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