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Update app.py
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app.py
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@@ -6,11 +6,8 @@ import librosa
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import json
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model_id = "cawoylel/windanam_mms-1b-tts_v2"
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# model = BetterTransformer.transform(model)
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pipe = pipeline("automatic-speech-recognition", model=model_id) #, tokenizer=processor.tokenizer, feature_extractor=processor.feature_extractor)
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def transcribe(audio_file_mic=None, audio_file_upload=None):
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if audio_file_mic:
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@@ -25,7 +22,18 @@ def transcribe(audio_file_mic=None, audio_file_upload=None):
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if sample_rate != 16000:
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speech = librosa.resample(speech, orig_sr=sample_rate, target_sr=16000)
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description = '''Automatic Speech Recognition with [MMS](https://ai.facebook.com/blog/multilingual-model-speech-recognition/) (Massively Multilingual Speech) by Meta.
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import json
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model_id = "cawoylel/windanam_mms-1b-tts_v2"
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processor = AutoProcessor.from_pretrained(model_id)
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model = Wav2Vec2ForCTC.from_pretrained(model_id)
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def transcribe(audio_file_mic=None, audio_file_upload=None):
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if audio_file_mic:
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if sample_rate != 16000:
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speech = librosa.resample(speech, orig_sr=sample_rate, target_sr=16000)
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# Keep the same model in memory and simply switch out the language adapters by calling load_adapter() for the model and set_target_lang() for the tokenizer
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processor.tokenizer.set_target_lang("ful")
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model.load_adapter(language_code)
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inputs = processor(speech, sampling_rate=16_000, return_tensors="pt")
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with torch.no_grad():
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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transcription = processor.decode(ids)
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return transcription
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description = '''Automatic Speech Recognition with [MMS](https://ai.facebook.com/blog/multilingual-model-speech-recognition/) (Massively Multilingual Speech) by Meta.
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