Crate lanczos_resampler 

Lanczos resampler

An audio resampler that uses Lanczos filter as an alternative to traditional windowed sinc filters. The main advantage of such approach is small number of coefficients required to store the filter state; this results in small memory footprint and high performance.

Features

Small memory footprint

The library doesn’t use memory allocation by default, and resampler’s internal state occupies less than a hundred bytes.

High performance

Thanks to small kernel size the processing time of a typical audio chunk is very fast (below 100 μs on a typical laptop).

Robustness

When you’re resampling from N Hz to M Hz, for each N input samples you will get exactly M output samples, provided that the output has enough space¹. This results in predictable audio stream playback and simplifies time synchronization between different streams (e.g. video and audio).

JS-compatible

This library can be used in web browsers and in general in any JS engine that supports WASM. All of the abovementioned features are inherent to both Rust and WASM versions of the library.

Usage

Kernel parameters

This library uses Lanczos kernel approximated by 2N - 1 points and defined on interval [-A; A]. The kernel is interpolated using cubic Hermite splines with second-order finite differences at spline endpoints. The output is clamped to [-1; 1].

The recommended parameters are N = 16, A = 3. Using A = 2 might improve performance a little bit. Using larger N will techincally improve precision, but precision isn’t a good metric for audio signal. With N = 16 the kernel fits into exactly 64 B (the size of a cache line).

Interleaved vs. non-interleaved format

Non-interleaved format means that audio samples for each channel are stored in separate arrays. To resample such data you need to call resample for each channel individually.

Interleaved format on the other hand means that samples for each channel are stored in a single array using frames; a frame is a sequence of samples, one sample for each channel. To resample such data you need to call resample only once.

Usually resampling interleaved data is much faster than processing each channel individually because a CPU can process such data efficiently with SIMD instructions.

Rust

Resampling audio stream in chunks

use lanczos_resampler::ChunkedResampler;

let n = 1024;
let chunk = vec![0.1; n];
let mut resampler = ChunkedResampler::new(44100, 48000);
let mut output: Vec<f32> = Vec::with_capacity(resampler.max_num_output_frames(n));
let num_processed = resampler.resample(&chunk[..], &mut output);
assert_eq!(n, num_processed);

Resampling the whole audio track

use lanczos_resampler::WholeResampler;

let n = 1024;
let track = vec![0.1; n];
let output_len = lanczos_resampler::num_output_frames(n, 44100, 48000);
let mut output = vec![0.0; output_len];
let resampler = WholeResampler::new();
let mut output_slice = &mut output[..];
let num_processed = resampler.resample_into(&track[..], &mut output_slice);
assert_eq!(n, num_processed);
assert!(output_slice.is_empty());

JS

Installation

npm install lanczos-resampler

Resampling audio stream in chunks

import { ChunkedResampler } from 'lanczos-resampler';

const resampler = new ChunkedResampler(44100, 48000);
const input = new Float32Array(1024);
input.fill(0.1);
const output = new Float32Array(resampler.maxNumOutputFrames(input.length));
const { numRead, numWritten } = resampler.resample(input, output);
assert.equal(input.length, numRead);

Resampling the whole audio track

import { WholeResampler, numOutputFrames } as lanczos from 'lanczos-resampler';

const input = new Float32Array(1024);
input.fill(0.1);
const outputLen = numOutputFrames(1024, 44100, 48000);
const output = new Float32Array(outputLen);
const resampler = new WholeResampler();
const { numRead, numWritten } = resampler.resampleInto(input, output);
assert.equal(input.length, numRead);
console.log(output)

Documentation

Rust: https://docs.rs/lanczos-resampler/latest/lanczos_resampler/

JS: https://igankevich.github.io/lanczos-resampler

No-std support

This crate supports no_std via libm. When std feature is enabled (the default), it uses built-in mathematical functions which are typically much faster than libm.

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1. Seriously, why other libraries don’t have this feature? ↩

Structs

BasicChunkedInterleavedResampleralloc

A resampler that processes audio input in chunks; the channels are interleaved with each other.

BasicChunkedResamplerNon-WebAssembly

A resampler that processes audio input in chunks.

BasicWholeResamplerNon-WebAssembly

A resampler that processes audio input as a whole.

Traits

Output

Resampler’s output, a buffer of samples.

WriteF32

A writable f32 audio sample.

Functions

checked_num_output_framesNon-WebAssembly

Calculates resampled length of the input for the given input/output sample rates.

num_output_framesNon-WebAssembly

Calculates resampled length of the input for the given input/output sample rates.

Type Aliases

ChunkedInterleavedResampleralloc

A BasicChunkedInterleavedResampler with default parameters: N = 16, A = 3.

ChunkedResamplerNon-WebAssembly

A BasicChunkedResampler with default parameters: N = 16, A = 3.

WholeResamplerNon-WebAssembly

A BasicWholeResampler with default parameters: N = 16, A = 3.