Model reference · open weights

wav2vec2-large-xlsr-53-french

Audio facebook Speech→text 1 build Open weights 2k dl/mo

wav2vec2-large-xlsr-53-french is an open-weight audio or speech model from Meta. wav2vec2-large-xlsr-53-french (BF16) weighs 1.3 GB; the smallest configuration that runs it is RTX 3060 12 GB.

For self-hosting, compare this build's requirements below with the GPU you plan to rent.

  • wav2vec2-large-xlsr-53-french is an automatic speech recognition model developed by Meta and published as facebook/wav2vec2-large-xlsr-53-french.
  • It is designed for processing French audio and is released under the apache-2.0 license.
  • The model card reports a word error rate of 25.2 percent on the Common Voice FR test set.

Summary of the facebook/wav2vec2-large-xlsr-53-french model card, 2026-10-01

What it is

Released byMeta
Released2022-03-02
VRAM1.3 GB for the weights

What it runs on

Memory and cards for wav2vec2-large-xlsr-53-french (BF16)

1.3 GBweights, file size
1.6 GBruntime overhead
CardOne streamMemory
RTX 3060 12 GBfits11.6 GB
RTX 4060 Ti 16 GBfits15.4 GB
RTX 3090 24 GBfits23.4 GB
RTX 4090 24 GBfits23.4 GB
RTX 5090 32 GBfits31.0 GB
L40S 48 GBfits44.0 GB
A100 80 GBfits78.2 GB
H100 80 GBfits78.1 GB
RTX PRO 6000 Blackwell 96 GBfits93.8 GB
DGX Spark (GB10) 128 GB unifiedfits107 GB
H200 141 GBfits138 GB
B200 180 GBfits176 GB

From the model card

What Meta says about wav2vec2-large-xlsr-53-french

Read the model card

Evaluation on Common Voice FR Test

import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys

model_name = "facebook/wav2vec2-large-xlsr-53-french"
device = "cuda"

chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"]'  # noqa: W605

model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(model_name)

ds = load_dataset("common_voice", "fr", split="test", data_dir="./cv-corpus-6.1-2020-12-11")

resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000)

def map_to_array(batch):
    speech, _ = torchaudio.load(batch["path"])
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch

ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
    pred_ids = torch.argmax(logits, dim=-1)
    batch["predicted"] = processor.batch_decode(pred_ids)
    batch["target"] = batch["sentence"]
    return batch

result = ds.map(map_to_pred, batched=True, batch_size=16, remove_columns=list(ds.features.keys()))

wer = load_metric("wer")

print(wer.compute(predictions=result["predicted"], references=result["target"]))

Result: 25.2 %

Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.

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