greek_lsr_1 / .ipynb_checkpoints /README-checkpoint.md

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	---
	language: {el}
	datasets:
	- common_voice
	metrics:
	- wer
	tags:
	- audio
	- automatic-speech-recognition
	- speech
	- xlsr-fine-tuning-week
	license: apache-2.0
	model-index:
	- name: {Greek XLSR Wav2Vec2 Large 53}
	results:
	- task:
	name: Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: Common Voice {el}
	type: common_voice
	args: {el}
	metrics:
	- name: Test WER
	type: wer
	value: {1.00}
	---

	# Wav2Vec2-Large-XLSR-53-{Greek}

	Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on {Greek} using the [Common Voice](https://huggingface.co/datasets/common_voice), ... and ... dataset{s}. #TODO: replace {language} with your language, e.g. French and eventually add more datasets that were used and eventually remove common voice if model was not trained on common voice
	When using this model, make sure that your speech input is sampled at 16kHz.

	## Usage

	The model can be used directly (without a language model) as follows:

	```python
	import torch
	import torchaudio
	from datasets import load_dataset
	from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
	test_dataset = load_dataset("common_voice", "{el}", split="test[:2%]") #TODO: replace {lang_id} in your language code here. Make sure the code is one of the ISO codes of [this](https://huggingface.co/languages) site.
	processor = Wav2Vec2Processor.from_pretrained("{skylord/greek_lsr_1}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, e.g. `elgeish/wav2vec2-large-xlsr-53-arabic`
	model = Wav2Vec2ForCTC.from_pretrained("{skylord/greek_lsr_1}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, e.g. `elgeish/wav2vec2-large-xlsr-53-arabic`
	resampler = torchaudio.transforms.Resample(48_000, 16_000)
	# Preprocessing the datasets.
	# We need to read the aduio files as arrays
	def speech_file_to_array_fn(batch):
	speech_array, sampling_rate = torchaudio.load(batch["path"])
	batch["speech"] = resampler(speech_array).squeeze().numpy()
	return batch
	test_dataset = test_dataset.map(speech_file_to_array_fn)
	inputs = processor(test_dataset["speech"][:2], 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)
	print("Prediction:", processor.batch_decode(predicted_ids))
	print("Reference:", test_dataset["sentence"][:2])
	```


	## Evaluation

	The model can be evaluated as follows on the {Greek} test data of Common Voice. # TODO: replace #TODO: replace language with your {language}, e.g. French


	```python
	import torch
	import torchaudio
	from datasets import load_dataset, load_metric
	from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
	import re
	test_dataset = load_dataset("common_voice", "{el}", split="test") #TODO: replace {lang_id} in your language code here. Make sure the code is one of the ISO codes of [this](https://huggingface.co/languages) site.
	wer = load_metric("wer")
	processor = Wav2Vec2Processor.from_pretrained("{skylord/greek_lsr_1}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, e.g. `elgeish/wav2vec2-large-xlsr-53-arabic`
	model = Wav2Vec2ForCTC.from_pretrained("{skylord/greek_lsr_1}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, e.g. `elgeish/wav2vec2-large-xlsr-53-arabic`
	model.to("cuda")
	chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\“]' # TODO: adapt this list to include all special characters you removed from the data
	resampler = torchaudio.transforms.Resample(48_000, 16_000)
	# Preprocessing the datasets.
	# We need to read the aduio files as arrays
	def speech_file_to_array_fn(batch):
	batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
	speech_array, sampling_rate = torchaudio.load(batch["path"])
	batch["speech"] = resampler(speech_array).squeeze().numpy()
	return batch
	test_dataset = test_dataset.map(speech_file_to_array_fn)
	# Preprocessing the datasets.
	# We need to read the aduio files as arrays
	def evaluate(batch):
	inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
	with torch.no_grad():
	logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
	pred_ids = torch.argmax(logits, dim=-1)
	batch["pred_strings"] = processor.batch_decode(pred_ids)
	return batch
	result = test_dataset.map(evaluate, batched=True, batch_size=8)
	print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
	```

	Test Result: 1.00 % # TODO: write output of print here. IMPORTANT: Please remember to also replace {wer_result_on_test} at the top of with this value here. tags.


	## Training

	The Common Voice `train`, `validation`, and ... datasets were used for training as well as ... and ... # TODO: adapt to state all the datasets that were used for training.

	The script used for training can be found [here](...) # TODO: fill in a link to your training script here. If you trained your model in a colab, simply fill in the link here. If you trained the model locally, it would be great if you could upload the training script on github and paste the link here.