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  Fine-tuned [facebook/wav2vec2-xls-r-1b](https://huggingface.co/facebook/wav2vec2-xls-r-1b) on Russian using the [Common Voice 8](https://huggingface.co/datasets/mozilla-foundation/common_voice_8_0).
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  When using this model, make sure that your speech input is sampled at 16kHz.
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- This model has been fine-tuned thanks to the GPU credits generously given by the [OVHcloud](https://www.ovhcloud.com/en/public-cloud/ai-training/) :)
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- The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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  ## Evaluation Commands
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  Fine-tuned [facebook/wav2vec2-xls-r-1b](https://huggingface.co/facebook/wav2vec2-xls-r-1b) on Russian using the [Common Voice 8](https://huggingface.co/datasets/mozilla-foundation/common_voice_8_0).
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  When using this model, make sure that your speech input is sampled at 16kHz.
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+ This model has been fine-tuned by the [HuggingSound](https://github.com/jonatasgrosman/huggingsound) tool, and thanks to the GPU credits generously given by the [OVHcloud](https://www.ovhcloud.com/en/public-cloud/ai-training/) :)
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+ ## Usage
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+
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+ Using the [HuggingSound](https://github.com/jonatasgrosman/huggingsound) library:
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+
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+ ```python
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+ from huggingsound import SpeechRecognitionModel
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+
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+ model = SpeechRecognitionModel("jonatasgrosman/wav2vec2-xls-r-1b-russian")
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+ audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
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+
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+ transcriptions = model.transcribe(audio_paths)
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+ ```
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+
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+ Writing your own inference script:
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+
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+ ```python
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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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+
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+ LANG_ID = "ru"
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+ MODEL_ID = "jonatasgrosman/wav2vec2-xls-r-1b-russian"
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+ SAMPLES = 10
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+
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+ test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
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+
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+ processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
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+ model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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+
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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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+
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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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+
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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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+
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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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+ ```
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  ## Evaluation Commands
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