Wav2Vec2 Base 960h
facebook/wav2vec2-base-960h
published Mar 2022 · updated Nov 2022
Wav2Vec2 Base 960h is an automatic speech recognition model that transcribes English speech audio into text, fine-tuned on 960 hours of LibriSpeech.
specs
| Task | Automatic Speech Recognition |
| Architecture | Wav2Vec2 Base |
| Training Data | 960 hours of LibriSpeech |
| Sampling Rate | 16 kHz |
about this model
Facebook Wav2Vec2-Base-960h is an automatic speech recognition (ASR) model that transcribes 16 kHz sampled speech audio into text. It is a self-supervised model pre-trained on raw audio and then fine-tuned on 960 hours of LibriSpeech data, using a contrastive task defined over quantized latent representations. This approach achieves competitive word error rates while being conceptually simpler than prior semi-supervised methods.
Key Capabilities
- State-of-the-art performance on LibriSpeech: 3.4% WER on the clean test set and 8.6% WER on the more challenging "other" test set.
- Robust to limited labeled data: the underlying wav2vec 2.0 framework (for which this model is a fine-tuned variant) can outperform previous methods using 100× less labeled data, as demonstrated in the original paper.
- Designed for 16 kHz input audio; the model directly processes raw waveforms without feature engineering.
Benchmark Results on LibriSpeech
| Test Set | Word Error Rate (WER) |
|---|---|
| Clean | 3.4% |
| Other | 8.6% |
These numbers reflect the Base-960h variant fine-tuned on all labeled LibriSpeech data. The larger model variant (paper reference) achieves lower WER (1.8%/3.3%) but uses a different architecture.
Background
The model is part of Facebook AI’s wav2vec 2.0 series (Baevski et al., 2020, arXiv:2006.11477). It learns powerful speech representations from raw audio alone, then fine-tunes on transcribed speech. Gigarouter hosts this model as a managed, OpenAI-compatible API — no local installation or framework handling required.
best for
- ·Transcribing English speech from audio files
- ·Real-time speech recognition for voice assistants
- ·Automated transcription of meetings or lectures
FAQ
The model expects mono audio sampled at 16 kHz. It can process raw audio arrays or audio files.
On LibriSpeech test sets, the model achieves 3.4% WER on clean speech and 8.6% WER on other (noisy) speech.
Send audio data to the gigarouter OpenAI-compatible endpoint with your API key. The model will return a text transcription.
It is a Wav2Vec2 Base model with a Transformer encoder and a CTC head for transcription.
The model outputs a text string containing the transcribed speech.
We're benchmarking and onboarding Wav2Vec2 Base 960h as a hosted, OpenAI-compatible API. Sign in for free credit and be ready when it lands, or tell us you want it and we'll prioritize it.