resampler

Resampler is a small, zero-dependency crate for high-quality audio resampling between common sample rates. It provides both FFT-based and FIR-based resamplers optimized for different use cases.

Usage Examples

FFT-Based Resampler (Highest Quality)

use resampler::{ResamplerFft, SampleRate};

// Create a stereo resampler (2 channels) from 44.1 kHz to 48 kHz.
let mut resampler = ResamplerFft::new(2, SampleRate::Hz44100, SampleRate::Hz48000);

// Get required buffer sizes (already includes all channels).
let input_size = resampler.chunk_size_input();
let output_size = resampler.chunk_size_output();

// Create input and output buffers (interleaved format: [L0, R0, L1, R1, ...]).
let input = vec![0.0f32; input_size];
let mut output = vec![0.0f32; output_size];

resampler.resample(&input, &mut output).unwrap();

FIR-Based Resampler (Low Latency, Streaming)

use resampler::{Attenuation, Latency, ResamplerFir, SampleRate};

// Create a stereo resampler with configurable latency (16, 32, or 64 samples).
let mut resampler = ResamplerFir::new(
    2,
    SampleRate::Hz48000,
    SampleRate::Hz44100,
    Latency::Sample64,
    Attenuation::Db90,
);

// Streaming API - accepts arbitrary input buffer sizes.
let input = vec![0.0f32; 512];
let mut output = vec![0.0f32; resampler.buffer_size_output()];

let (consumed, produced) = resampler.resample(&input, &mut output).unwrap();
println!("Consumed {consumed} samples, produced {produced} samples");

Choosing a Resampler

Both resamplers provide good quality, but are optimized for different use cases:

Use ResamplerFft when:

• You need the absolute highest quality
• Latency is not a concern
• Processing pre-recorded audio files

Use ResamplerFir when:

• You need low latency (real-time audio)
• You can live with a slower rolloff
• Working with streaming data

FFT-Based Implementation

The resampler uses an FFT-based overlap-add algorithm with Kaiser windowing for high-quality audio resampling. Key technical details:

• Custom mixed-radix FFT with the Stockham Autosort algorithm.
• SIMD optimizations: All butterflies have SSE2, SSE4.2, AVX+FMA, and ARM NEON implementations.
• Stopband attenuation of -100 dB using the Kaiser windows function.
• Latency around 256 samples.

FIR-Based Implementation

The FIR resampler uses a polyphase filter with linear interpolation for high-quality audio resampling with low latency. Key technical details:

• Polyphase decomposition: 1024 phases with linear interpolation between phases.
• SIMD optimizations: Convolution kernels optimized with SSE2, SSE4.2, AVX+FMA, AVX-512, and ARM NEON.
• Configurable filter length: 32, 64, or 128 taps (16, 32, or 64 samples latency).
• Adjustable rolloff and stopband attenuation.
• Streaming API: Accepts arbitrary input buffer sizes for flexible real-time processing.

Performance

Both resamplers include SIMD optimizations with runtime CPU feature detection for maximum performance and compatibility.

But for up to 25% better performance on x86_64, compile with target-cpu=x86-64-v3 (enables AVX2, FMA, and other optimizations).

Overall the SIMD for x86_64 have four levels implemented, targeting four possible CPU generations that build up on each other:

• x86-64-v1: 128-bit SSE2 (around 2003-2004)
• x86-64-v2: 128-bit SSE4.2 (around 2008-2011)
• x86-64-v3: 256-bit AVX+FMA (around 2013-2015)
• x86-64-v4: 512-bit AVX-512 (around 2017-2022)

no-std Compatibility

The library supports no-std environments with alloc. To use the library in a no-std environment, enable the no_std feature:

[dependencies]
resampler = { version = "0.2", features = ["no_std"] }

Behavior Differences

When the no_std feature is enabled:

• Caching: The library will not cache FFT and FIR objects globally to shorten resampler creation time and lower overall memory consumption for multiple resamplers.

• No runtime detection of SIMD functionality. You need to activate SIMD via compile time target features.

The default build (without no_std feature) has zero dependencies and uses the standard library for optimal performance and memory efficiency through global caching.

Quality Analysis

The following spectrograms demonstrate the high-quality output of the resampler across different conversion scenarios:

44.1 kHz → 48 kHz Conversion With FFT Resampler

44.1 kHz → 48 kHz Conversion With FIR Resampler

Alternatives

Other high-quality audio resampling libraries in Rust are:

• Rubato: The overlap-add resampling approach used in this library is based on Rubato's implementation.

License

Licensed under either of

• Apache License, Version 2.0, (LICENSE-APACHE or http://www.apache.org/licenses/LICENSE-2.0)
• MIT license (LICENSE-MIT or http://opensource.org/licenses/MIT)

at your option.

Contribution

Unless you explicitly state otherwise, any contribution intentionally submitted for inclusion in the work by you, as defined in the Apache-2.0 license, shall be dual licensed as above, without any additional terms or conditions.