alt.freeweights.speech / s2t-small-librispeech-asr
From: facebook <huggingface.co/facebook>Newsgroups: alt.freeweights.speechSubject:

s2t-small-librispeech-asr

Date: 2026-11-15Commit: 256e91aSigned: 2DC28CC7B26A283B ✓

Automatic speech recognition model with 30M parameters. Safetensors weights, 366 MB in all. Published by facebook under MIT.

30M params366MBs2t-small-librispeech-asrsafetensorspermissive
FactsParams30MSize366 MBFormatssafetensorsTaskautomatic speech recognitionLicenseMITFrom sourcepermissive
$ fw get facebook/s2t-small-librispeech-asr
# manifest sha256 9c3d02fd4394… · 366 MB · 12 files
$ fw verify ./s2t-small-librispeech-asr facebook/s2t-small-librispeech-asr
From sourceHosted (signed)facebook/s2t-small-librispeech-asrLast check: 2026-11-15 02:51 UTC

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Works with

Overview

Size
366 MB
Weights
118 MB (safetensors)
Parameters
30M
Formats
safetensors
Task
automatic speech recognition
License
MIT (permissive)
Files
12, each with a sha256
Publisher
facebook
Added
2026-11-15
Section
alt.freeweights.speech
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Model card

From the publisher at commit 256e91a. Shown as plain text with links; images open at the publisher.

S2T-SMALL-LIBRISPEECH-ASR#

s2t-small-librispeech-asr is a Speech to Text Transformer (S2T) model trained for automatic speech recognition (ASR). The S2T model was proposed in this paper and released in this repository

Model description#

S2T is an end-to-end sequence-to-sequence transformer model. It is trained with standard autoregressive cross-entropy loss and generates the transcripts autoregressively.

Intended uses & limitations#

This model can be used for end-to-end speech recognition (ASR). See the model hub to look for other S2T checkpoints.

How to use#

As this a standard sequence to sequence transformer model, you can use the generate method to generate the transcripts by passing the speech features to the model.

Note: The Speech2TextProcessor object uses torchaudio to extract the filter bank features. Make sure to install the torchaudio package before running this example.

Note: The feature extractor depends on torchaudio and the tokenizer depends on sentencepiece so be sure to install those packages before running the examples.

You could either install those as extra speech dependancies with pip install transformers"[speech, sentencepiece]" or install the packages seperatly with pip install torchaudio sentencepiece.

import torch
from transformers import Speech2TextProcessor, Speech2TextForConditionalGeneration
from datasets import load_dataset

model = Speech2TextForConditionalGeneration.from_pretrained("facebook/s2t-small-librispeech-asr")
processor = Speech2TextProcessor.from_pretrained("facebook/s2t-small-librispeech-asr")

ds = load_dataset(
    "patrickvonplaten/librispeech_asr_dummy",
    "clean",
    split="validation"
)

input_features = processor(
    ds[0]["audio"]["array"],
    sampling_rate=16_000,
    return_tensors="pt"
).input_features  # Batch size 1
generated_ids = model.generate(input_features=input_features)

transcription = processor.batch_decode(generated_ids)
Evaluation on LibriSpeech Test#

The following script shows how to evaluate this model on the LibriSpeech "clean" and "other" test dataset.

from datasets import load_dataset
from evaluate import load
from transformers import Speech2TextForConditionalGeneration, Speech2TextProcessor

librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")  # change to "other" for other test dataset
wer = load("wer")

model = Speech2TextForConditionalGeneration.from_pretrained("facebook/s2t-small-librispeech-asr").to("cuda")
processor = Speech2TextProcessor.from_pretrained("facebook/s2t-small-librispeech-asr", do_upper_case=True)

def map_to_pred(batch):
    features = processor(batch["audio"]["array"], sampling_rate=16000, padding=True, return_tensors="pt")
    input_features = features.input_features.to("cuda")
    attention_mask = features.attention_mask.to("cuda")

    gen_tokens = model.generate(input_features=input_features, attention_mask=attention_mask)
    batch["transcription"] = processor.batch_decode(gen_tokens, skip_special_tokens=True)[0]
    return batch

result = librispeech_eval.map(map_to_pred, remove_columns=["audio"])

print("WER:", wer.compute(predictions=result["transcription"], references=result["text"]))

Result (WER):

"clean""other"
4.39.0

Training data#

The S2T-SMALL-LIBRISPEECH-ASR is trained on LibriSpeech ASR Corpus, a dataset consisting of approximately 1000 hours of 16kHz read English speech.

Training procedure#

Preprocessing#

The speech data is pre-processed by extracting Kaldi-compliant 80-channel log mel-filter bank features automatically from WAV/FLAC audio files via PyKaldi or torchaudio. Further utterance-level CMVN (cepstral mean and variance normalization) is applied to each example.

The texts are lowercased and tokenized using SentencePiece and a vocabulary size of 10,000.

Training#

The model is trained with standard autoregressive cross-entropy loss and using SpecAugment. The encoder receives speech features, and the decoder generates the transcripts autoregressively.

BibTeX entry and citation info#
@inproceedings{wang2020fairseqs2t,
  title = {fairseq S2T: Fast Speech-to-Text Modeling with fairseq},
  author = {Changhan Wang and Yun Tang and Xutai Ma and Anne Wu and Dmytro Okhonko and Juan Pino},
  booktitle = {Proceedings of the 2020 Conference of the Asian Chapter of the Association for Computational Linguistics (AACL): System Demonstrations},
  year = {2020},
}

Files

12 files · 366 MBWeights 118 MB (safetensors). Sizes and sha256 from the signed manifest.
Hide all 12 files with their sha256
FileSizesha256
.gitattributes744 Bf888dc7c230287cd68af7034aa9fc2e2d6663fb7b3e70faa153caf93877914c3
README.md6.00 KB76c6a96a989fa9742ebc79ef4e95d586489196a761482475819d7bcb1477b427
config.json1.14 KB908bc8e3f28d83ed24ffd1fd83fb113d18b388820699f1b08318b62d0729dd77
generation_config.json233 Baed76366507333ddbb8bd49960f23c82fe6446b3319a46a54befdb45324ccf61
model.safetensors118 MB7d2b5fd0d9072cf00d3599363653a91f725d24e50a6b9ece8e4cb0837ba1969f
preprocessor_config.json242 B6cd3ccc0cd0741e4e9fcdbb11ad634ff6b773710e9dfdb6d7a0e09a755a12836
pytorch_model.bin118 MB95be85b800e626fa6063bf30bd40874b3a426fc12b0393b7046546e470fcc535
sentencepiece.bpe.model417 KB052a168787a9160b4b2ba54e4995e9600298812c34191ca3f70cea51cd4f5c1e
special_tokens_map.json85 Bbb7068de1150661a10b55f9e4b12a0e77af8bf91f5e45e1b58afaf1d0e17f675
tf_model.h5129 MB2faac8e98aaa3808196dab18955801120c7aab1c6d4d17ea788fefd1cd37aaa8
tokenizer_config.json456 B81db5f6f936011645eec56e0656748b95ef993c031dfa7230efd7a990bbe4b3b
vocab.json235 KBd101b11eba9f37e488b6f41a48b3bdac8a6112e8bac86ea4b437a40dd344ccf1
VersionsEach time the pinned upstream commit or file list changed in this repo. Newest first.
Upstream commitDateContents
256e91a603 pinned now2026-11-15 02:51 UTC12 files · 366 MB

Use it

$ fw get facebook/s2t-small-librispeech-asr
# manifest sha256 9c3d02fd4394… · 366 MB · 12 files
$ fw verify ./s2t-small-librispeech-asr facebook/s2t-small-librispeech-asr
fw.pyChecks the signature and every sha256, tries each source in order, resumes.
$ python3 fw.py get https://freeweights.allternit.com/manifests/facebook/s2t-small-librispeech-asr.json --dir ./s2t-small-librispeech-asr $ python3 fw.py verify ./s2t-small-librispeech-asr ./facebook__s2t-small-librispeech-asr.json
hfThe Hugging Face CLI with one setting. Same repo id.
$ HF_ENDPOINT=https://freeweights.allternit.com hf download facebook/s2t-small-librispeech-asr --revision 256e91a603c013ae09aa1e4c09cfc3f8acdf8d19 --local-dir ./s2t-small-librispeech-asr
transformersSpeech recognition with the transformers pipeline.
import os os.environ["HF_ENDPOINT"] = "https://freeweights.allternit.com" from transformers import pipeline asr = pipeline("automatic-speech-recognition", model="facebook/s2t-small-librispeech-asr", revision="256e91a603c013ae09aa1e4c09cfc3f8acdf8d19") print(asr("audio.wav")["text"])

License

Name
MIT License
Id
MIT
Class
Permissive
Redistribution
yes
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License as declared by the publisher. No person read it. 2026-11-15
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README.md
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76c6a96a989fa9742ebc79ef4e95d586489196a761482475819d7bcb1477b427
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Provenance

From: facebook <huggingface.co/facebook>Newsgroups: alt.freeweights.speechSubject: s2t-small-librispeech-asrDate: 2026-11-15Commit: 256e91aSigned: 2DC28CC7B26A283B ✓
Upstream
facebook/s2t-small-librispeech-asr
Pinned commit
256e91a603c013ae09aa1e4c09cfc3f8acdf8d19
Manifest
s2t-small-librispeech-asr.json · .minisig
Manifest sha256
9c3d02fd4394edafbfd91ecd0665e9051686d72193bdad6b7278864025651947
Signature
minisign, Ed25519, key 2DC28CC7B26A283B
Created
2026-11-15 02:51 UTC
Generator
freeweights-manifest/0.1.0
Takedowns
None.

How to verify

$ curl -O https://freeweights.allternit.com/manifests/facebook/s2t-small-librispeech-asr.json $ curl -O https://freeweights.allternit.com/manifests/facebook/s2t-small-librispeech-asr.json.minisig $ minisign -Vm s2t-small-librispeech-asr.json -P RWQ7KGqyx4zCLXtUFOqvfv8L3g+8fpKJ1FsuxPyOYMF+4KufbUxKq1Oq $ shasum -a 256 <file> # compare with the Files tab

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Signed manifest
minisign, key 2DC28CC7B26A283B. s2t-small-librispeech-asr.json. Provenance. Takedown history: none.
From: MIT <huggingface.co/MIT>Subject: ast-finetuned-audioset-10-10-0.4593
From: onnx-community <huggingface.co/onnx-community>Subject: ast-finetuned-audioset-10-10-0.4593-ONNX
From: Edge0 <huggingface.co/Edge0>Subject: Audio8-TTS-Preview-0.6b
From: Edge0 <huggingface.co/Edge0>Subject: Audio8-TTS-Realistic-Air
From: facebook <huggingface.co/facebook>Subject: audiobox-aesthetics
From: mlx-community <huggingface.co/mlx-community>Subject: chatterbox-4bit