AIAny
AI Audio2022
Icon for item

FunASR

Bundles ASR, voice activity detection, punctuation, and speaker diarization into one pipeline, with pretrained models like Paraformer and SenseVoice. SenseVoice runs ~17x realtime on CPU; also ships streaming ASR and an OpenAI-compatible API.

Introduction

Whisper trained everyone to reach for a single, giant, do-everything model. FunASR bets the opposite way: ship a toolkit of smaller, purpose-built components — a VAD model, an ASR model, a punctuation model, a diarization model — that you compose into one pipeline. That modularity is the whole point, and it is why a 234M-parameter SenseVoice can outrun a much larger general model on Mandarin while still running on a plain CPU.

What Sets It Apart
  • Paraformer is non-autoregressive: it decodes a whole utterance in parallel rather than token-by-token, so latency stays flat as audio grows — the source of the headline realtime multiples.
  • The models were trained on Alibaba's industrial corpora (tens of thousands of hours of annotated Mandarin), so accuracy on Chinese accents and telephony-quality audio tends to beat Western-trained baselines.
  • You pick the model that matches the job rather than accepting one compromise: SenseVoice for emotion plus multilingual on CPU, Paraformer for timestamped Mandarin, larger LLM-based models for 30+ languages.
  • Production plumbing is built in — streaming over WebSocket and an OpenAI-compatible endpoint — so swapping it in for a hosted API is closer to a URL change than a rewrite.
Who It's For

Great fit if you process Mandarin or mixed Chinese-English audio at scale, need word-level timestamps or speaker labels, or want CPU-only deployment without a GPU bill. Look elsewhere if your audio is mostly English, where Whisper's ecosystem and tooling run deeper, or if you want one drop-in model and would rather not reason about which components to chain. Documentation skews toward Chinese, so non-Chinese readers should expect some translation friction.

Information

  • Websitegithub.com
  • OrganizationsAlibaba DAMO Academy, ModelScope
  • AuthorsAlibaba DAMO Academy, Northwestern Polytechnical University (NWPU), China Telecom, RapidAI, AIHealthX, XVERSE, Community contributors
  • Published date2022/11/24

More Items

GitHub
AI Audio2025

Provides text-to-speech synthesis and one-shot voice cloning via compact ONNX models and a larger KittenTTS 2 speech model. Offers CPU-first inference (25–506 MB models), 47 built-in voices, multilingual output, expression tags, streaming, and a Python SDK.

Hugging Face
AI Audio2026

Generates directed, multi-character, long-form speech from text with natural-language directions and voice cloning. Built on a 35B MoE backbone (3B active per token) with a codebook multi-token-prediction decoder to enable multi-turn revision, infilling edits, streaming, and expressive control.

GitHub
AI Audio2025

Generates full songs from lyrics and style prompts by first writing an editable melody-and-chord score and then rendering vocals and accompaniment. Unifies symbolic planning and audio synthesis in one pipeline, enabling readable score inspection, zero‑shot covers, and agent-driven editing.