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README.md
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## Model description
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##
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##
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## Training procedure
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## Model description
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The architecture is the same as [openai/whisper-medium](https://huggingface.co/openai/whisper-medium).
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## Training and evaluation data
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The model was trained on the Common Voice 11.0 dataset (`train+validation+other splits`) and the Romanian speech synthesis corpus, and was tested on the `test` split of the Common Voice 11.0 dataset.
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## Usage
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Inference with 🤗 Pipeline
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```python
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import torch
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from datasets import load_dataset
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from transformers import pipeline
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device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
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# Load pipeline
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pipe = pipeline("automatic-speech-recognition", model="gigant/whisper-medium-romanian", device=device)
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# NB: set forced_decoder_ids for generation utils
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pipe.model.config.forced_decoder_ids = pipe.tokenizer.get_decoder_prompt_ids(language="ro", task="transcribe")
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# Load data
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ds_mcv_test = load_dataset("mozilla-foundation/common_voice_11_0", "ro", split="test", streaming=True)
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test_segment = next(iter(ds_mcv_test))
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waveform = test_segment["audio"]
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# NB: decoding option
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# limit the maximum number of generated tokens to 225
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pipe.model.config.max_length = 225 + 1
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# sampling
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# pipe.model.config.do_sample = True
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# beam search
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# pipe.model.config.num_beams = 5
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# return
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# pipe.model.config.return_dict_in_generate = True
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# pipe.model.config.output_scores = True
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# pipe.model.config.num_return_sequences = 5
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# Run
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generated_sentences = pipe(waveform)["text"]
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```
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Inference with 🤗 low-level APIs
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```python
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import torch
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import torchaudio
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from datasets import load_dataset
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from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
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device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
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# Load model
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model = AutoModelForSpeechSeq2Seq.from_pretrained("gigant/whisper-medium-romanian").to(device)
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processor = AutoProcessor.from_pretrained("gigant/whisper-medium-romanian", language="romanian", task="transcribe")
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# NB: set forced_decoder_ids for generation utils
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model.config.forced_decoder_ids = processor.get_decoder_prompt_ids(language="ro", task="transcribe")
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# 16_000
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model_sample_rate = processor.feature_extractor.sampling_rate
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# Load data
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ds_mcv_test = load_dataset("mozilla-foundation/common_voice_11_0", "ro", split="test", streaming=True)
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test_segment = next(iter(ds_mcv_test))
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waveform = torch.from_numpy(test_segment["audio"]["array"])
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sample_rate = test_segment["audio"]["sampling_rate"]
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# Resample
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if sample_rate != model_sample_rate:
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resampler = torchaudio.transforms.Resample(sample_rate, model_sample_rate)
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waveform = resampler(waveform)
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# Get feat
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inputs = processor(waveform, sampling_rate=model_sample_rate, return_tensors="pt")
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input_features = inputs.input_features
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input_features = input_features.to(device)
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# Generate
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generated_ids = model.generate(inputs=input_features, max_new_tokens=225) # greedy
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# generated_ids = model.generate(inputs=input_features, max_new_tokens=225, num_beams=5) # beam search
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# Detokenize
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generated_sentences = processor.batch_decode(generated_ids, skip_special_tokens=True)[0]
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# Normalise predicted sentences if necessary
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```
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The code was adapted from [bofenghuang/deprecated-whisper-large-v2-cv11-french-punct-plus](https://huggingface.co/bofenghuang/deprecated-whisper-large-v2-cv11-french-punct-plus).
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## Training procedure
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