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facebook_wav2vec2-base-960h

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language: en datasets:

  • librispeech_asr tags:
  • audio
  • automatic-speech-recognition
  • hf-asr-leaderboard license: apache-2.0 widget:
  • example_title: Librispeech sample 1 src: https://cdn-media.huggingface.co/speech_samples/sample1.flac
  • example_title: Librispeech sample 2 src: https://cdn-media.huggingface.co/speech_samples/sample2.flac model-index:
  • name: wav2vec2-base-960h results:
    • task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: LibriSpeech (clean) type: librispeech_asr config: clean split: test args: language: en metrics:
      • name: Test WER type: wer value: 3.4
    • task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: LibriSpeech (other) type: librispeech_asr config: other split: test args: language: en metrics:
      • name: Test WER type: wer value: 8.6

Wav2Vec2-Base-960h

Facebook's Wav2Vec2

The base model pretrained and fine-tuned on 960 hours of Librispeech on 16kHz sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.

Paper

Authors: Alexei Baevski, Henry Zhou, Abdelrahman Mohamed, Michael Auli

Abstract

We show for the first time that learning powerful representations from speech audio alone followed by fine-tuning on transcribed speech can outperform the best semi-supervised methods while being conceptually simpler. wav2vec 2.0 masks the speech input in the latent space and solves a contrastive task defined over a quantization of the latent representations which are jointly learned. Experiments using all labeled data of Librispeech achieve 1.8/3.3 WER on the clean/other test sets. When lowering the amount of labeled data to one hour, wav2vec 2.0 outperforms the previous state of the art on the 100 hour subset while using 100 times less labeled data. Using just ten minutes of labeled data and pre-training on 53k hours of unlabeled data still achieves 4.8/8.2 WER. This demonstrates the feasibility of speech recognition with limited amounts of labeled data.

The original model can be found under https://github.com/pytorch/fairseq/tree/master/examples/wav2vec#wav2vec-20.

Usage

To transcribe audio files the model can be used as a standalone acoustic model as follows:

 from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
 from datasets import load_dataset
 import torch
 
 # load model and tokenizer
 processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h")
 model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h")
     
 # load dummy dataset and read soundfiles
 ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
 
 # tokenize
 input_values = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest").input_values  # Batch size 1
 
 # retrieve logits
 logits = model(input_values).logits
 
 # take argmax and decode
 predicted_ids = torch.argmax(logits, dim=-1)
 transcription = processor.batch_decode(predicted_ids)

Evaluation

This code snippet shows how to evaluate facebook/wav2vec2-base-960h on LibriSpeech's "clean" and "other" test data.

from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import torch
from jiwer import wer


librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")

model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h").to("cuda")
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h")

def map_to_pred(batch):
    input_values = processor(batch["audio"]["array"], return_tensors="pt", padding="longest").input_values
    with torch.no_grad():
        logits = model(input_values.to("cuda")).logits

    predicted_ids = torch.argmax(logits, dim=-1)
    transcription = processor.batch_decode(predicted_ids)
    batch["transcription"] = transcription
    return batch

result = librispeech_eval.map(map_to_pred, batched=True, batch_size=1, remove_columns=["audio"])

print("WER:", wer(result["text"], result["transcription"]))

Result (WER):

"clean" "other"
3.4 8.6

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README.md4.3 KB (4,431 B)c7fe2047d7ac9b9816848c657b2a492ee95b264bfc52a0adfa64e446800f9944c6cf9b90da3127aba548860ea5714516ec8c18c7
config.json1.6 KB (1,596 B)8ca9cc7496e145e37d09cec17d0c3bf9b8523c8ed3ec255c063d9f95057b553b19c20135b259875834a4fe9deb218a6be25b4cf3
feature_extractor_config.json158 B (158 B)52fdd74dc06f40033506e402269fbde5e7adc21dd3de0c797bf9b65f90bc65c30cb7b303ebeda341f6fc80af33628c4b26b95632
model.safetensors360.1 MB (377,607,901 B)48aa0a8ea67ac86db70f12c016323e7c12d56c5f8aa76ab2243c81747a1f832954586bc566090c83a0ac167df6f31f0fa917d74a
preprocessor_config.json159 B (159 B)3f24dc078fcba55ee1d417a413847ead40c093a3b225d617c025463b9e157e06afea8b90dc7078fc70b013c533328423e0486b4a
pytorch_model.bin360.2 MB (377,667,514 B)decd44471e17d2fe1fef40c3301144fe200ce324c34f9827b034a1b9141dbf6f652f8a60eda61cdf5771c9e05bfa99033c92cd96
special_tokens_map.json85 B (85 B)25bc39604f72700b3b8e10bd69bb2f227157edd1bb7068de1150661a10b55f9e4b12a0e77af8bf91f5e45e1b58afaf1d0e17f675
tokenizer_config.json163 B (163 B)978a15a96dbb2d23e2afbc70137cae6c5ce38c8ddc790594f5bc351a4311c6624f40acd95850d4aaf2a5cb3c656c9b610720b608
vocab.json291 B (291 B)88181b954aa14df68be9b444b3c36585f3078c0a19727f8944fe6459fc3f240ae2c198395b740f6a029bd23e06656266b83bcf64

Cite this release

Canonical URL
https://aiseedbank.org/models/facebook_wav2vec2-base-960h/
Slug
facebook_wav2vec2-base-960h
Infohash
43edb3f6e6ac552395fbf7b43298cc35f85a7f3d
License
apache-2.0
Signing key fingerprint
85a3b32c3712427b

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Upstream repositoryfacebook/wav2vec2-base-960h
Revision (pinned)22aad52d435eb6dbaf354bdad9b0da84ce7d6156
Fetched at2026-09-03T22:58:20Z
License at fetchapache-2.0
Snapshot toolhuggingface · seedbank 0.1.0

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✓ verified · rehash-vs-hf-metadata at 2026-09-03T22:58:28Z

apache-2.0720.3 MB (755,282,298 bytes)transformerspytorchsafetensorswav2vec2automatic-speech-recognitionaudiohf-asr-leaderboardmodel-indexeval-resultsendpoints_compatible2 languages (tf, en)paper: 2006.11477