AIAny
AI Audio2019
Icon for item

NVIDIA NeMo Speech

Provides a toolkit and codebase for building, training, and deploying speech and multimodal models — Automatic Speech Recognition, Text-to-Speech, and speech-aware LLMs — with modular neural components and pre-trained checkpoints for PyTorch. Supports streaming/low-latency inference, multi-language models, and optional compiled kernels for acceleration.

Introduction

NeMo Speech arrives at a moment when speech-first applications demand both research flexibility and production reliability: low-latency streaming ASR, high-quality TTS, and speech-aware LLMs share engineering needs (scalable training, checkpoints, and optimized inference) that many single-purpose repos do not address. NeMo packages those building blocks into a single, extensible toolkit so teams can move from prototype to deployable systems without reimplementing core components.

What Sets It Apart
  • Modular neural components designed for PyTorch workflows — you can mix ASR, TTS, speaker and audio-processing modules into custom pipelines, which reduces reimplementation when experimenting with architectures.
  • Production-oriented checkpoints and demos — the repo maintains pretrained models and example inference paths aimed at streaming and low-latency modes, so teams can evaluate real-world latency/accuracy trade-offs quickly.
  • Flexible installation and optimized backends — works as a pip/uv package or container-based stack, and optionally benefits from compiled kernels (Transformer Engine, FlashAttention, grouped-GEMM/MoE) for throughput on supported NVIDIA GPUs. The Automodel backend can run without compiled dependencies for easier experimentation.
  • Focused on audio + multimodal LLMs — recent development emphasizes speech LLMs and unified speech stacks (ASR→LLM→TTS) rather than a general-purpose multimodal library.
Who It's For and Trade-offs

Great fit if you need a production-minded speech toolkit with research-friendly modularity: teams building streaming ASR services, custom TTS voices, or speech-capable LLM integrations will find ready checkpoints and integration patterns. The project also suits PyTorch-centric labs that want optional, high-performance GPU kernels for large-scale training and low-latency inference.

Look elsewhere if you want a minimal, single-file library or only CPU-only inference: NeMo is a substantial codebase with many optional compiled components and is optimized for GPU-enabled workflows. If you require a tiny runtime for edge devices with no GPU support, a lightweight inference-only runtime may be a better choice.

Overall, NeMo Speech is a pragmatic middle ground: it reduces duplicated engineering across ASR/TTS/voice-LLM projects while exposing the performance knobs teams need when moving from research to production.

More Items

Hugging Face

Provides 3,451 hours (2,051,810 clips) of AI‑generated 48 kHz Turkish speech with transcripts, spoken forms and per‑clip voice descriptions for TTS and ASR development. Includes 2,752 designed voices and is licensed CC BY 4.0 / CC BY‑SA 4.0 (attribution to PatientDesk AI required).

Hugging Face
AI Audio2026

Performs speaker diarization (who spoke when) for live and recorded audio using an open-weight, 100M-parameter streaming-capable model that supports up to eight anonymous speaker channels, overlapping speech, chunked processing, and configurable latency for ASR integration.

Hugging Face
AI Audio2026

Performs low-latency streaming speech-to-text, emitting one token per selectable 80/120/160 ms clock with configurable transcription delay and a 30s rolling KV cache for unlimited 24/7 transcription. Bilingual (zh/en) and includes semantic VAD.