sf-limiter

sf-limiter (short for “straightforward limiter”) is a look-ahead brick-wall audio limiter with a dependency-free Rust core and optional Python bindings for NumPy.

It applies one linked gain value to every channel in a frame, preserving the relative balance between channels.

Non-streaming: The current API processes a complete audio buffer offline. Each call starts with a fresh gain envelope, so limiter state does not carry across chunks or successive calls. The output has the same shape and length as the input, while the limiter uses future samples equal to its attack-time look-ahead.

Note: The limiter algorithm itself was not AI-generated. Codex was used only to help with API design, documentation, and packaging.

Python 3.11+

Install the package from PyPI:

python -m pip install sf-limiter

Limit a mono NumPy array:

import numpy as np
import sf_limiter

audio = np.array([0.0, 0.5, 3.0, -4.0, 0.25], dtype=np.float64)
limited, frame_gains = sf_limiter.limit(audio, sample_rate=48_000)

assert limited.dtype == np.float32
assert np.max(np.abs(limited), initial=0.0) <= 1.0

The one-shot limit function accepts these keyword parameters:

  • threshold_dBFS=0.0 (dBFS)
  • attack_ms=5.0
  • hold_ms=15.0
  • release_ms=40.0
  • axis=-1

For repeated use, configure a limiter object once:

limiter = sf_limiter.SFLimiter(
    48_000,
    threshold_dBFS=-1.0,
    attack_ms=5.0,
    hold_ms=15.0,
    release_ms=40.0,
)
limited, frame_gains = limiter.process(audio)

limiter.threshold_dBFS returns the configured dBFS value, while limiter.threshold returns the corresponding linear amplitude.

Input may be a one-dimensional mono array or a two-dimensional multichannel array. The last axis is interpreted as frames by default, so the usual shape is (channels, frames). Pass axis=0 for (frames, channels). The input is converted to float32 without being mutated; the returned audio is a new float32 array, and frame_gains contains one value per frame.

Each call starts from a neutral gain envelope. Non-finite samples and invalid configuration values raise ValueError.

Rust

The core Rust API accepts flat f32 samples in either of these layouts:

  • Frame-interleaved: each frame contains one sample per channel. Use process_interleaved or process_interleaved_inplace.
  • Channel-planar: all frames of the first channel are followed by all frames of the next channel. Use process_planar or process_planar_inplace.

For frame-interleaved audio:

use sf_limiter::SFLimiter;

let input = [0.0, 0.5, 3.0, -4.0, 0.25];
let mut limiter = SFLimiter::with_default(48_000)?;
let output = limiter.process_interleaved(&input, 1)?;

assert!(output.audio.iter().all(|sample| sample.abs() <= 1.0));
# Ok::<(), sf_limiter::LimiterError>(())

Use process_interleaved_inplace to reuse the input allocation. Both methods return one linked gain value per frame and reset the envelope on every call.

For channel-planar audio:

let mut planar = [0.0, 0.5, 3.0, -4.0, 0.25, -0.25];
let mut limiter = SFLimiter::with_default(48_000)?;
let frame_gains = limiter.process_planar_inplace(&mut planar, 2)?;

assert_eq!(frame_gains.len(), 3);
# Ok::<(), sf_limiter::LimiterError>(())

The Python binding always uses the planar core path. Default two-dimensional axis=-1 input is already planar and is processed directly; (frames, channels) input is transposed to planar layout for processing and then restored to its original layout for the returned array. The interleaved core path remains available to Rust callers.

Ceiling guarantee

For finite input, a valid channel count, and a finite threshold_dBFS no greater than 0.0 dBFS, every returned discrete sample is finite and has an absolute value no greater than the corresponding linear ceiling (10 ** (threshold_dBFS / 20)). The test suite checks this with large impulses, high-level deterministic noise, mono input, and linked multichannel input.

This is a sample-peak limiter. It does not oversample to detect reconstructed inter-sample (true-peak) excursions.

Design reference

The limiter design was informed by Geraint Luff's “Designing a straightforward limiter” (Signalsmith Audio, 2022). In particular, this implementation follows the article's look-ahead structure: a moving minimum of permissible gain, an exponential release, and finite-length cascaded box-filter smoothing.

The Rust implementation was extracted from limiter.rs in thesia and adapted into a standalone crate with a dependency-free core.

Development

Run the Rust tests:

cargo test

Create the Python environment with the test dependencies, build the extension, and run its tests:

uv sync
uv run pytest -q

Compare the Python API performance with numpy-audio-limiter:

uv sync --group benchmark
uv run --group benchmark python benchmarks/compare_numpy_audio_limiter.py

The benchmark uses contiguous channel-planar arrays shaped (channels, frames) for both implementations and measures reusable and one-shot sf_limiter calls separately. Use --help to select durations, channel counts, timing repetitions, and limiter settings.

TODO

  • Refine the Rust API
  • Add a streaming API
  • Publish the crate to crates.io
  • Implement an optional true-peak limiter with oversampling

Project links

Data verified by PyPI on Aug 17, 2026
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  • Released:
Newer release available (0.1.1)

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Avatar for sytronik from gravatar.com sytronik

Credits

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Author: Sytronik

GitHub Statistics

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License expression

MIT
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Requires

Python >=3.11

Tags

audio dsp limiter

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