AIAny
AI Audio2026
Icon for item

Confucius4-R2T2

Provides low-latency, true-streaming automatic speech recognition that emits append-only committed transcripts to avoid partial-result rollbacks. Supports configurable decoding chunks (80 ms–2 s), optimized for Chinese and English, and offers vLLM and transformers backends for real-time deployment.

Introduction

Most streaming ASR systems trade off user-facing stability for accuracy by emitting provisional tokens that are later revised, causing distracting text flicker in live captions and agent pipelines. R2T2 flips that trade-off: it exposes only stable prefixes so incremental outputs are final, while keeping latency in the 200–600 ms range and accuracy close to offline recognition.

Key Capabilities
  • Longest Stable Prefix (LSP) streaming: the model is trained and decoded to decide when a token sequence is safe to commit, eliminating revisions in the incremental transcript — so clients can consume emitted text immediately without reconciliation logic.
  • Fine-grained chunking: configurable decoding chunks from 80 ms to 2 s let you choose latency vs. accuracy operating points; typical setups report average latencies of ~200–600 ms.
  • Robust accuracy and multilingual support: built as a Qwen3-ASR fine-tune with techniques like stable-prefix data and token-level audio segmentation; optimized for Chinese and English and usable on many other languages.
  • Production-friendly integrations: ships examples for vLLM and Hugging Face transformers backends, a WebSocket server for multi-client streaming, and support for context/hotword prompts.
Who it's for and trade-offs

Great fit if you need real-time, append-only transcripts for live captioning, simultaneous translation pipelines, LLM agents that act on streaming text, or downstream NLP systems that cannot tolerate text revisions. The project includes server/client examples and a vLLM-based backend for high-throughput inference.

Look elsewhere if you require fully permissive model weights for arbitrary model training or commercial deployment at very large scale without reviewing license terms: the code is Apache-2.0 but the model weights are distributed under NetEase Model Use License Agreement, which imposes reuse and commercial-scale restrictions. Also, production deployment expects GPU/vLLM or compatible runtime; CPU-only deployments will have higher latency.

Where it fits

R2T2 positions itself between pseudo-streaming systems (which revise partial outputs) and offline ASR: it aims to deliver offline-grade accuracy while providing stable, low-latency incremental outputs for real-time applications.

Implementation notes

The project includes Docker configurations, a streaming WebSocket server, and Python examples for both offline and streaming inference. Typical usage configures chunk size, unfixed token window, and optional hotword/context prompts; example runtimes use vLLM for throughput and audio.cpp/gguf derivatives for lightweight local inference.

Information

  • Websitehuggingface.co
  • OrganizationsNetEase Youdao
  • Published date2026/09/10

Categories

More Items

Hugging Face
AI Model2026

Generates speech from text and can clone a voice from a reference recording without fine-tuning. Its 0.6B-parameter core supports 23 languages, though quality varies substantially by language.

GitHub
AI Audio2026

Turns speech into text in whichever app is active: hold a shortcut, speak, and release. Transcription runs locally after the speech model is downloaded, keeping routine dictation on your device.

Hugging Face
AI Model2026

Generates speech from text and can clone a voice from a reference recording without fine-tuning. The 601M-parameter model supports 23 languages, though output quality varies by language.