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README.md
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HIYACCENT: An Improved Nigerian-Accented Speech Recognition System Based on Contrastive Learning
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The global objective of this research was to develop a more robust model for the Nigerian English Speakers whose English pronunciations are heavily affected by their mother tongue.
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HIYACCENT: An Improved Nigerian-Accented Speech Recognition System Based on Contrastive Learning
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The global objective of this research was to develop a more robust model for the Nigerian English Speakers whose English pronunciations are heavily affected by their mother tongue. For this, the Wav2Vec-HIYACCENT model was proposed which introduced a new layer to the Novel Facebook Wav2vec to capture the disparity between the baseline model and Nigerian English Speeches. A CTC loss was also inserted on top of the model which adds flexibility to the speech-text alignment. This resulted in over 20% improvement in the performance for NAE.T
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Fine-tuned facebook/wav2vec2-large on English using the UISpeech Corpus. When using this model, make sure that your speech input is sampled at 16kHz.
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The script used for training can be found here: https://github.com/amceejay/HIYACCENT-NE-Speech-Recognition-System
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Usage
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The model can be used directly (without a language model) as follows...
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Using the ASRecognition library:
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from asrecognition import ASREngine
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asr = ASREngine("fr", model_path="codeceejay/HIYACCENT_Wav2Vec2")
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audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
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transcriptions = asr.transcribe(audio_paths)
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Writing your own inference speech:
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import torch
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import librosa
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from datasets import load_dataset
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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LANG_ID = "en"
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MODEL_ID = "codeceejay/HIYACCENT_Wav2Vec2"
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SAMPLES = 10
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#You can use common_voice/timit or Nigerian Accented Speeches can also be found here: https://openslr.org/70/
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test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
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processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
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model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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# Preprocessing the datasets.
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# We need to read the audio files as arrays
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def speech_file_to_array_fn(batch):
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speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
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batch["speech"] = speech_array
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batch["sentence"] = batch["sentence"].upper()
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return batch
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test_dataset = test_dataset.map(speech_file_to_array_fn)
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inputs = processor(test_dataset["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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predicted_sentences = processor.batch_decode(predicted_ids)
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for i, predicted_sentence in enumerate(predicted_sentences):
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print("-" * 100)
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print("Reference:", test_dataset[i]["sentence"])
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print("Prediction:", predicted_sentence)
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