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import torch |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline |
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from transformers.pipelines.audio_utils import ffmpeg_read |
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import gradio as gr |
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import librosa |
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MODEL_NAME = "EwoutLagendijk/whisper-small-indonesian" |
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BATCH_SIZE = 8 |
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device = 0 if torch.cuda.is_available() else "cpu" |
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model_name = "EwoutLagendijk/whisper-small-indonesian" |
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model = AutoModelForSpeechSeq2Seq.from_pretrained(model_name) |
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processor = AutoProcessor.from_pretrained(model_name) |
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model.config.forced_decoder_ids = processor.get_decoder_prompt_ids(language="id", task="transcribe") |
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def transcribe_speech(filepath): |
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audio, sampling_rate = librosa.load(filepath, sr=16000) |
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chunk_duration = 30 |
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chunk_samples = chunk_duration * sampling_rate |
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transcription = [] |
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for i in range(0, len(audio), chunk_samples): |
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chunk = audio[i:i + chunk_samples] |
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inputs = processor(audio=chunk, sampling_rate=16000, return_tensors="pt").input_features |
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generated_ids = model.generate( |
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inputs, |
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max_new_tokens=444, |
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forced_decoder_ids=processor.get_decoder_prompt_ids(language="id", task="transcribe"), |
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return_timestamps = True |
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) |
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chunk_transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)[0] |
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transcription.append(chunk_transcription) |
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return " ".join(transcription) |
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demo = gr.Blocks() |
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mic_transcribe = gr.Interface( |
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fn=transcribe_speech, |
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inputs=gr.Audio(sources="microphone", type="filepath"), |
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outputs=gr.components.Textbox(), |
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) |
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file_transcribe = gr.Interface( |
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fn=transcribe_speech, |
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inputs=gr.Audio(sources="upload", type="filepath"), |
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outputs=gr.components.Textbox(), |
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) |
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with demo: |
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gr.TabbedInterface([mic_transcribe, file_transcribe], ["Transcribe Microphone", "Transcribe Audio File"]) |
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demo.launch(debug=True) |