Disclaimer and Requirements
This model is a clone of jonatasgrosman/wav2vec2-large-xlsr-53-english compressed using ZipNN. Compressed losslessly to 88% its original size, ZipNN saved ~0.2GB in storage and potentially ~4PB in data transfer monthly.
Requirement
In order to use the model, ZipNN is necessary:
pip install zipnn
Use This Model
# Use a pipeline as a high-level helper
from transformers import pipeline
from zipnn import zipnn_hf
zipnn_hf()
pipe = pipeline("automatic-speech-recognition", model="royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed")
# Load model directly
from transformers import AutoProcessor, AutoModelForCTC
from zipnn import zipnn_hf
zipnn_hf()
processor = AutoProcessor.from_pretrained("royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed")
model = AutoModelForCTC.from_pretrained("royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed")
ZipNN
ZipNN also allows you to seemlessly save local disk space in your cache after the model is downloaded.
To compress the cached model, simply run:
python zipnn_compress_path.py safetensors --model royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed --hf_cache
The model will be decompressed automatically and safely as long as zipnn_hf()
is added at the top of the file like in the example above.
To decompress manualy, simply run:
python zipnn_decompress_path.py --model royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed --hf_cache
Fine-tuned XLSR-53 large model for speech recognition in English
Fine-tuned facebook/wav2vec2-large-xlsr-53 on English using the train and validation splits of Common Voice 6.1. When using this model, make sure that your speech input is sampled at 16kHz.
This model has been fine-tuned thanks to the GPU credits generously given by the OVHcloud :)
The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint
Usage
The model can be used directly (without a language model) as follows...
Using the HuggingSound library:
from huggingsound import SpeechRecognitionModel
from zipnn import zipnn_hf
zipnn_hf()
model = SpeechRecognitionModel("royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed")
audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
transcriptions = model.transcribe(audio_paths)
Writing your own inference script:
import torch
import librosa
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
from zipnn import zipnn_hf
zipnn_hf()
LANG_ID = "en"
MODEL_ID = "royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed"
SAMPLES = 10
test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
batch["speech"] = speech_array
batch["sentence"] = batch["sentence"].upper()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
predicted_ids = torch.argmax(logits, dim=-1)
predicted_sentences = processor.batch_decode(predicted_ids)
for i, predicted_sentence in enumerate(predicted_sentences):
print("-" * 100)
print("Reference:", test_dataset[i]["sentence"])
print("Prediction:", predicted_sentence)
Reference | Prediction |
---|---|
"SHE'LL BE ALL RIGHT." | SHE'LL BE ALL RIGHT |
SIX | SIX |
"ALL'S WELL THAT ENDS WELL." | ALL AS WELL THAT ENDS WELL |
DO YOU MEAN IT? | DO YOU MEAN IT |
THE NEW PATCH IS LESS INVASIVE THAN THE OLD ONE, BUT STILL CAUSES REGRESSIONS. | THE NEW PATCH IS LESS INVASIVE THAN THE OLD ONE BUT STILL CAUSES REGRESSION |
HOW IS MOZILLA GOING TO HANDLE AMBIGUITIES LIKE QUEUE AND CUE? | HOW IS MOSLILLAR GOING TO HANDLE ANDBEWOOTH HIS LIKE Q AND Q |
"I GUESS YOU MUST THINK I'M KINDA BATTY." | RUSTIAN WASTIN PAN ONTE BATTLY |
NO ONE NEAR THE REMOTE MACHINE YOU COULD RING? | NO ONE NEAR THE REMOTE MACHINE YOU COULD RING |
SAUCE FOR THE GOOSE IS SAUCE FOR THE GANDER. | SAUCE FOR THE GUICE IS SAUCE FOR THE GONDER |
GROVES STARTED WRITING SONGS WHEN SHE WAS FOUR YEARS OLD. | GRAFS STARTED WRITING SONGS WHEN SHE WAS FOUR YEARS OLD |
Evaluation
- To evaluate on
mozilla-foundation/common_voice_6_0
with splittest
python eval.py --model_id royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed --dataset mozilla-foundation/common_voice_6_0 --config en --split test
- To evaluate on
speech-recognition-community-v2/dev_data
python eval.py --model_id royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed --dataset speech-recognition-community-v2/dev_data --config en --split validation --chunk_length_s 5.0 --stride_length_s 1.0
Citation
If you want to cite this model you can use this:
@misc{grosman2021xlsr53-large-english,
title={Fine-tuned {XLSR}-53 large model for speech recognition in {E}nglish},
author={Grosman, Jonatas},
howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-large-xlsr-53-english}},
year={2021}
}
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Datasets used to train royleibov/wav2vec2-large-xlsr-53-english-ZipNN-Compressed
Evaluation results
- Test WER on Common Voice enself-reported19.060
- Test CER on Common Voice enself-reported7.690
- Test WER (+LM) on Common Voice enself-reported14.810
- Test CER (+LM) on Common Voice enself-reported6.840
- Dev WER on Robust Speech Event - Dev Dataself-reported27.720
- Dev CER on Robust Speech Event - Dev Dataself-reported11.650
- Dev WER (+LM) on Robust Speech Event - Dev Dataself-reported20.850
- Dev CER (+LM) on Robust Speech Event - Dev Dataself-reported11.010