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@@ -58,79 +58,28 @@ The architecture is the same as [openai/whisper-medium](https://huggingface.co/o
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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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-
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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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-
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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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-
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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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-
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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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-
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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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-
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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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-
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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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-
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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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  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 🤗 transformers
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  ```python
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+ from transformers import WhisperProcessor, WhisperForConditionalGeneration
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+ from datasets import Audio, load_dataset
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  import torch
 
 
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+ # load model and processor
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+ processor = WhisperProcessor.from_pretrained("gigant/whisper-medium-romanian")
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+ model = WhisperForConditionalGeneration.from_pretrained("gigant/whisper-medium-romanian")
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+
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+ # load dummy dataset and read soundfiles
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+ ds = load_dataset("common_voice", "ro", split="test", streaming=True)
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+ ds = ds.cast_column("audio", Audio(sampling_rate=16_000))
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+ input_speech = next(iter(ds))["audio"]["array"]
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+ model.config.forced_decoder_ids = processor.get_decoder_prompt_ids(language = "ro", task = "transcribe")
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+ input_features = processor(input_speech, return_tensors="pt", sampling_rate=16_000).input_features
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+ predicted_ids = model.generate(input_features, max_length=448)
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+ # transcription = processor.batch_decode(predicted_ids)
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+ transcription = processor.batch_decode(predicted_ids, skip_special_tokens = True)
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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  ```
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+ The code was adapted from [openai/whisper-medium](https://huggingface.co/openai/whisper-medium).
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  ## Training procedure
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