audiosronnx 0.4.0a1

Pure-ONNX multi-engine audio super-resolution / bandwidth extension (upscale speech to 48 kHz)

audiosronnx

Pure-ONNX, multi-engine audio super-resolution / bandwidth extension. Upscale low-sample-rate or low-bandwidth speech — 8 kHz telephony, muffled recordings, low-fidelity TTS — to a clean 48 kHz signal. Inference runs entirely on onnxruntime; there is no torch at runtime.

audiosronnx is the super-resolution sibling of the TigreGotico pure-ONNX speech libraries (vadonnx, voiceclonnx, speakeronnx) and mirrors their conventions: a single load_sr(engine=...) entry point, an engine registry, a HuggingFace-backed model resolver with an XDG cache, and a small CLI.

Install

pip install audiosronnx

Runtime dependencies: numpy, onnxruntime, scipy, soundfile, huggingface_hub. ONNX models are downloaded on first use from the Hugging Face Hub and cached under ~/.local/share/audiosronnx.

Engines

Engine	Input SR	Output SR	Size	CPU speed	License	Status
lavasr (default)	8–48 kHz	48 kHz	~52 MB	~50× realtime	Apache-2.0	shipped
novasr	16 kHz	48 kHz	~0.2 MB	~1000× realtime	Apache-2.0	shipped
hifiganbwe	any	48 kHz	~4 MB	fast	MIT	shipped
apbwe	any (12 kHz band)	48 kHz	~120 MB	moderate	MIT	shipped
sidon	16 kHz	48 kHz	~410 MB	~0.6× realtime (CPU)	MIT	shipped
callenhancer	8–16 kHz	48 kHz	~3 GB fp32 / ~1.3 GB int8	~0.3–0.4× realtime (CPU)	CC-BY-NC-4.0	shipped

• lavasr — a Vocos-based bandwidth-extension model with a Linkwitz-Riley spectral merge that preserves the original low band, plus an optional UL-UNAS denoiser. It accepts any input rate from 8 to 48 kHz. All spectral DSP (STFT, ISTFT, mel filterbank, resampling, merge) runs in numpy/scipy, so no torch.stft ever enters an ONNX graph.
• novasr — a tiny conv1d / BigVGAN-snake generator that upsamples 16 kHz to 48 kHz in a single time-domain pass. Extremely fast and memory-light, at lower fidelity than lavasr; useful for on-device enhancement and quick dataset restoration.
• hifiganbwe — HiFi-GAN+ (Su et al., ICASSP 2021): bandlimited (kaiser) interpolation to 48 kHz followed by a non-causal WaveNet that reconstructs the high band. Time-domain, ~1M params, no spectral front-end. Accepts any input rate.
• apbwe — AP-BWE (Lu et al.): dual-ConvNeXt amplitude-and-phase prediction in the STFT domain; the strongest log-spectral-distance accuracy of the shipped engines. The packaged checkpoint is the 12 kHz→48 kHz model.
• sidon — Sidon (SARULab-Speech): full speech restoration, not just bandwidth extension. An 8-layer w2v-BERT 2.0 feature predictor (LoRA-adapted to denoise SSL representations) feeds a DAC vocoder that resynthesises clean 48 kHz audio from degraded 16 kHz input. The SeamlessM4T log-mel front-end runs in numpy; the feature extractor ships int8-quantized. It is the heaviest shipped engine — best on GPU, but runs on CPU at roughly 0.6× realtime, which is fine for offline dataset cleansing.
• callenhancer — CallEnhancer (Scicom-intl): call-centre / telephony restoration. Same two-graph shape as sidon but with the full 24-layer w2v-BERT 2.0 feature predictor (LoRA-merged) driving a 188M-param DAC vocoder, tuned on narrowband, codec'd call audio (G.711/GSM, 8–16 kHz). Heavier and slower than sidon, but the strongest restoration of the shipped engines on real telephony. Runs a single length-invariant pass by default (chunk_seconds>0 windows very long calls with a crossfade). The ONNX weights are CC-BY-NC-4.0 (research / non-commercial) — the model itself, not audio you restore with it.

lavasr/novasr derive from the LavaSR/NovaSR projects by Yatharth Sharma (LavaSR, NovaSR, Apache-2.0); hifiganbwe from brentspell/hifi-gan-bwe (MIT); apbwe from yxlu-0102/AP-BWE (MIT); sidon from sarulab-speech/Sidon (MIT); callenhancer from Scicom-intl/CallEnhancer (CC-BY-NC-4.0). Every neural component runs through onnxruntime with all STFT/ISTFT/resampling kept in numpy/scipy — the ONNX graphs are validated against the original PyTorch models (HiFi-GAN+ end-to-end correlation 1.0000, AP-BWE 0.9998, per-graph max error ≤ 5e-4).

Precision (int8 quantization)

The two transformer-based engines ship an int8-quantized feature extractor to keep the download and CPU footprint manageable. Quantization is not free, and how much it costs depends on the model — so pick precision per engine, not by reflex:

Engine	fp32 FE	int8 FE	int8 vs fp32 (end-to-end)	Default	Recommendation
sidon (8-layer FE)	—	~410 MB	near-lossless	int8	int8 — the shallow encoder quantizes cleanly
callenhancer (24-layer FE)	~2.3 GB	~580 MB	corr 0.969, ~12 dB SNR (audible)	fp32	fp32 for fidelity; int8 only when size/speed matters more than quality

The deeper the encoder, the more per-layer int8 rounding error accumulates: Sidon's 8-layer extractor stays effectively lossless, but CallEnhancer's full 24-layer w2v-BERT loses roughly 12 dB SNR end-to-end — enough to hear. So callenhancer defaults to the fp32 feature extractor (weights ride in an external .onnx.data sidecar, fetched automatically alongside the graph); pass precision="int8" to trade fidelity for a ~4× smaller, faster download:

sr = load_sr("callenhancer")                    # fp32 FE (default, full fidelity)
sr = load_sr("callenhancer", precision="int8")  # ~580 MB, faster, audibly lossy

Every decoder ships fp32 (the DAC vocoder is small and quantizes poorly).

Quickstart

from audiosronnx import load_sr

sr = load_sr("lavasr")                      # default engine

# low-level primitive: numpy array or path in, (float32 mono, 48000) out
out, rate = sr.upscale("telephone_8k.wav")  # -> (np.ndarray float32, 48000)

# array input (you supply the sample rate)
import soundfile as sf
audio, in_sr = sf.read("input.wav")
out, rate = sr.upscale(audio, in_sr)

# file / directory helpers
sr.upscale_file("in.wav", "out_48k.wav")
sr.upscale_dir("clips_in/", "clips_out_48k/")

# the fast tiny engine instead
fast = load_sr("novasr")
fast.upscale_file("in_16k.wav", "out_48k.wav")

LavaSR options:

sr = load_sr("lavasr", denoise=True, cutoff_hz=6000)

CLI

audiosronnx list                              # list engines
audiosronnx probe input.wav                   # print format / duration
audiosronnx upscale in.wav out.wav            # default engine (lavasr)
audiosronnx upscale in.wav out.wav --engine novasr
audiosronnx upscale in.wav out.wav --denoise  # lavasr denoiser stage
audiosronnx upscale-dir clips_in/ clips_out/

API

• load_sr(engine="lavasr", *, providers=None, cache_dir=None, revision=None, **kwargs) -> SRModel
• SRModel.upscale(audio, sample_rate=None) -> (np.ndarray float32 mono, 48000) — audio is a path, raw int16 PCM bytes, or a numpy array. The low-level primitive.
• SRModel.upscale_file(in_path, out_path) -> str
• SRModel.upscale_dir(in_dir, out_dir) -> list[str]
• available_models(), get_engine(alias), register_engine(entry), ENGINE_REGISTRY

Custom / third-party engines

Subclass SRModel, implement _upscale_array(audio, sample_rate) -> np.ndarray (a 48 kHz mono float32 array), and register an EngineEntry:

from audiosronnx import SRModel, EngineEntry, register_engine

class MySR(SRModel):
    output_sample_rate = 48000
    def _upscale_array(self, audio, sample_rate):
        ...  # run your ONNX session, return a 48 kHz float32 array

register_engine(EngineEntry(alias="mysr", adapter_class=MySR, license="MIT"))

Model hosting

Exported ONNX weights are hosted on the Hugging Face Hub and pinned by revision:

• TigreGotico/audiosronnx-lavasr — backbone.onnx, spec_head.onnx, denoiser_core.onnx
• TigreGotico/audiosronnx-novasr — novasr.onnx
• TigreGotico/audiosronnx-hifiganbwe — hifiganbwe_wavenet.onnx
• TigreGotico/audiosronnx-apbwe — apbwe.onnx
• TigreGotico/audiosronnx-sidon — feature_extractor.int8.onnx, decoder.onnx

The export scripts under conversion/ reproduce these from the upstream PyTorch checkpoints and validate each ONNX graph against its PyTorch submodule (max absolute error).

Evaluated but not shipped

The audio super-resolution landscape was surveyed for other models to export. These were evaluated and are not shipped, for the reasons given:

Model	License	Reason not shipped
FLowHigh	MIT	Single-step flow matching, but depends on an external BigVGAN vocoder plus a mel/STFT front-end — a multi-component export rather than one clean graph.
AudioSR	MIT	~6 GB latent-diffusion model (VAE + LDM + vocoder, iterative sampler, ~0.6× realtime on GPU). Not CPU-runnable at usable latency; impractical to export.
resemble-enhance	MIT	Four-network 44.1 kHz restoration pipeline: UNet denoiser + IRMAE autoencoder + a CFM iterative ODE sampler (32–64 velocity-net evals per utterance) + a UnivNet vocoder. The vocoder is LVCNet — a location-variable convolution whose kernels are predicted per location and applied via einsum over unfolds, so it does not fold into a static ONNX graph. Speech restoration/denoising, not single-pass bandwidth extension (its hparams hard-assert 44.1 kHz). Same diffusion-style, multi-graph, dynamic-op class as AudioSR.
NU-Wave2	none	Diffusion (iterative sampler) and the repository ships no license file.
mdctGAN	unclear (NOASSERTION)	Unclear license, and its MDCT front-end relies on torch.fft, which exports to ONNX unreliably.
VoiceFixer / NVSR	MIT	Speech restoration rather than pure bandwidth extension; two-stage mel-predictor + neural vocoder, heavier and less focused than the shipped engines.

An engine is only shipped when it exports cleanly to ONNX, runs on CPU via onnxruntime, carries a permissive license, and produces verified 48 kHz output.

License

Apache-2.0. The shipped model weights are Apache-2.0 (LavaSR, NovaSR).