AIAny
AI Audio2019
Icon for item

NVIDIA NeMo

Build, fine-tune, and deploy speech AI on NVIDIA GPUs: ASR, text-to-speech, and speech LLMs in one PyTorch stack. Ships pretrained Parakeet/Canary recognition and Magpie TTS checkpoints; broader LLM/multimodal training now lives in v2.7.0.

Introduction

NeMo started life as NVIDIA's catch-all framework for training LLMs, multimodal models, and speech systems alike. Its more interesting recent move is the opposite of feature creep: the project narrowed, pushing broad LLM/multimodal training back to release v2.7.0 and reorienting the active codebase around speech and audio. That focus is the signal — this is now where NVIDIA ships its production speech recognition and synthesis models, not a generic trainer.

What Sets It Apart
  • The recognition lineup is state-of-the-art and open: Parakeet and Canary models cover 25 European languages, Canary-Qwen reports ~5.6% WER on the English Open ASR Leaderboard, and a streaming Nemotron checkpoint lets you slide along a latency-accuracy curve (roughly 80ms to 1s) instead of retraining.
  • TTS isn't an afterthought — Magpie speaks nine languages and a unified Parakeet model handles both offline and streaming inference with punctuation, so one model covers more of a real pipeline.
  • Speech LLMs like Nemotron VoiceChat target full-duplex, low-latency conversation, the hard part of voice agents that cascaded ASR-then-LLM-then-TTS stacks handle poorly.
  • Checkpoints ship on Hugging Face and NGC, so you start from a strong base rather than training from scratch.
Who It's For

Great fit if you're building voice agents, transcription, or TTS on NVIDIA hardware and want open, fine-tunable models with a real deployment path. Look elsewhere if you came for general LLM or multimodal pretraining — that now means pinning v2.7.0 — or if you run on non-NVIDIA accelerators or just want a hosted speech API rather than a framework you operate yourself.

Information

  • Websitegithub.com
  • AuthorsNVIDIA
  • Published date2019/08/05

More Items

GitHub
AI Audio2026

Zero-shot text-to-speech across 600+ languages with short-reference voice cloning and attribute-based voice design. Uses a diffusion-style discrete non-autoregressive architecture mapping text to multi-codebook acoustic tokens, supports phoneme/pinyin control and very low RTF.

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.