--- language: mt datasets: - common_voice tags: - audio - automatic-speech-recognition - maltese - xlrs-53-maltese - masri-project - malta - university-of-malta license: cc-by-4.0 widget: model-index: - name: wav2vec2-large-xlsr-53-maltese-64h results: - task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: Mozilla Common Voice 11.0 (Test) type: mozilla-foundation/common_voice_11_0 split: test args: language: mt metrics: - name: Test WER type: wer value: 1.57 - task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: Mozilla Common Voice 11.0 (Dev) type: mozilla-foundation/common_voice_11_0 split: dev args: language: mt metrics: - name: Test WER type: wer value: 1.40 --- # wav2vec2-large-xlsr-53-maltese-64h The "wav2vec2-large-xlsr-53-maltese-64h" is an acoustic model suitable for Automatic Speech Recognition in Maltese. It is the result of fine-tuning the model "facebook/wav2vec2-large-xlsr-53" with around 64 hours of Maltese data developed by the MASRI Project at the University of Malta between 2019 and 2021. Most of the data is available at the the MASRI Project homepage https://www.um.edu.mt/projects/masri/. The specific list of corpora used to fine-tune the model is: - MASRI-HEADSET v2 (6h39m) - MASRI-Farfield (9h37m) - MASRI-Booths (2h27m) - MASRI-MEP (1h17m) - MASRI-COMVO (7h29m) - MASRI-TUBE (13h17m) - MASRI-MERLIN (25h18m) *Not available at the MASRI Project homepage The fine-tuning process was perform during November (2022) in the servers of the Language and Voice Lab (https://lvl.ru.is/) at Reykjavík University (Iceland) by Carlos Daniel Hernández Mena. # Evaluation ```python import torch from transformers import Wav2Vec2Processor from transformers import Wav2Vec2ForCTC #Load the processor and model. MODEL_NAME="carlosdanielhernandezmena/wav2vec2-large-xlsr-53-maltese-64h" processor = Wav2Vec2Processor.from_pretrained(MODEL_NAME) model = Wav2Vec2ForCTC.from_pretrained(MODEL_NAME) #Load the dataset from datasets import load_dataset, load_metric, Audio ds=load_dataset("common_voice", "mt", split="test") #Normalize the transcriptions import re chars_to_ignore_regex = '[\\,\\?\\.\\!\\\;\\:\\"\\“\\%\\‘\\”\\�\\)\\(\\*)]' def remove_special_characters(batch): batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower() return batch ds = ds.map(remove_special_characters) #Downsample to 16kHz ds = ds.cast_column("audio", Audio(sampling_rate=16_000)) #Process the dataset def prepare_dataset(batch): audio = batch["audio"] #Batched output is "un-batched" to ensure mapping is correct batch["input_values"] = processor(audio["array"], sampling_rate=audio["sampling_rate"]).input_values[0] with processor.as_target_processor(): batch["labels"] = processor(batch["sentence"]).input_ids return batch ds = ds.map(prepare_dataset, remove_columns=ds.column_names,num_proc=1) #Define the evaluation metric import numpy as np wer_metric = load_metric("wer") def compute_metrics(pred): pred_logits = pred.predictions pred_ids = np.argmax(pred_logits, axis=-1) pred.label_ids[pred.label_ids == -100] = processor.tokenizer.pad_token_id pred_str = processor.batch_decode(pred_ids) #We do not want to group tokens when computing the metrics label_str = processor.batch_decode(pred.label_ids, group_tokens=False) wer = wer_metric.compute(predictions=pred_str, references=label_str) return {"wer": wer} #Do the evaluation (with batch_size=1) model = model.to(torch.device("cuda")) def map_to_result(batch): with torch.no_grad(): input_values = torch.tensor(batch["input_values"], device="cuda").unsqueeze(0) logits = model(input_values).logits pred_ids = torch.argmax(logits, dim=-1) batch["pred_str"] = processor.batch_decode(pred_ids)[0] batch["sentence"] = processor.decode(batch["labels"], group_tokens=False) return batch results = ds.map(map_to_result,remove_columns=ds.column_names) #Compute the overall WER now. print("Test WER: {:.3f}".format(wer_metric.compute(predictions=results["pred_str"], references=results["sentence"]))) ``` **Test Result**: 0.011 # BibTeX entry and citation info *When publishing results based on these models please refer to:* ```bibtex @misc{mena2022xlrs53maltese, title={Acoustic Model in Maltese: wav2vec2-large-xlsr-53-maltese-64h.}, author={Hernandez Mena, Carlos Daniel}, year={2022}, url={https://huggingface.co/carlosdanielhernandezmena/wav2vec2-large-xlsr-53-maltese-64h}, } ``` # Acknowledgements The MASRI Project is funded by the University of Malta Research Fund Awards. We want to thank to Merlin Publishers (Malta) for provinding the audiobooks used to create the MASRI-MERLIN Corpus. Special thanks to Jón Guðnason, head of the Language and Voice Lab for providing computational power to make this model possible. We also want to thank to the "Language Technology Programme for Icelandic 2019-2023" which is managed and coordinated by Almannarómur, and it is funded by the Icelandic Ministry of Education, Science and Culture.