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    #Rate conversion

    This package converts interleaved floating-point PCM without changing the declared playback sample rate. rate is input frames advanced per output frame: a value of 2 doubles pitch and halves duration. Supported rates are 0.25 through 4, with 1 through 8 channels.

    new(channels, rate) creates the converter. Feed complete frames to push, consume the returned output, then append the result of finish. No host I/O, native DSP library, or SoundTouch implementation is used. reset reuses the configured filter for an independent stream.

    For non-identity rates, the kernel is a normalized Blackman-windowed sinc. Its normalized cutoff is 0.9 / max(1, rate) and its input-frame radius is ceil(48 * max(1, rate)). A table of 513 phases is built once; runtime samples interpolate between adjacent coefficient rows. Downsampling lowers the cutoff and increases the support rather than aliasing high frequencies through a fixed interpolation filter. All channels use the same fractional position.

    The filter intentionally reserves a transition band below Nyquist; it is not a brick-wall filter. Lookahead is exposed by latency_frames. Output begins when the first center has its complete future support. finish extends the first and last samples, preserves DC at the edges, and emits exactly floor(input_frames / rate + 0.5) frames. Identity conversion is an immediate, bit-exact copy, including signed zero. The retained history after a push is bounded by twice the filter radius plus four frames; input and returned output arrays remain the caller's responsibility.

    The tests measure passband waveform error, at least 60 dB stopband rejection for specified 1.37x, 2x and 4x downsampling tones, impulse timing, reported lookahead, endpoint DC, signed-zero identity, invalid-input atomicity, final duration, per-channel isolation, and exact input-partition invariance. These are numerical checks of this rate converter, not perceptual quality claims for the complete time-stretch pipeline.

    Algorithm background: Julius O. Smith III, Windowed Sinc Interpolation. The implementation in this repository is original; no source code is copied from that reference or from SoundTouch.

    Processor

    source
    pub struct Processor {
      channels : Int
      rate : Double
      radius : Int
      coefficients : Array[Array[Double]]
      samples : Array[Double]
      base_frame : Int64
      input_frames : Int64
      output_frames : Int64
      next_frame : Int64
      fraction : Double
      finished : Bool
    }

    Continuous interleaved PCM rate conversion. Rate is input frames advanced per output frame: 2 halves duration and doubles pitch. All channels share the same phase and filter. Rate 1 passes samples through exactly.

    Processor::buffered_frames

    source
    fn Processor::buffered_frames(self : Processor) -> Int

    Input frames retained after a call, excluding returned output.

    Processor::finish

    source
    fn Processor::finish(self : Processor) -> Array[Double]

    Finish using endpoint extension. Final length is nearest(input_frames / rate), with half frames rounded upward. Repeated finish calls return empty.

    Processor::latency_frames

    source
    fn Processor::latency_frames(self : Processor) -> Int

    Lookahead in input frames. A shorter stream is emitted by finish.

    Processor::push

    source
    fn Processor::push(self : Processor, samples : Array[Double]) -> Array[Double] raise 
    AudioError

    Push complete frames. Invalid input leaves the processor unchanged. Output is independent of how the input is divided among calls.

    Processor::reset

    source
    fn Processor::reset(self : Processor) -> Unit

    Discard sample history and counters, retaining configuration and filters.

    new

    source
    fn new(channels : Int, rate : Double) -> Processor raise 
    AudioError

    Original Blackman-windowed sinc converter. Downsampling lowers the cutoff and widens the kernel in input frames, preserving its transition width.

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