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app.py
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import gradio as gr
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import librosa
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import torch
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from transformers import WhisperProcessor, WhisperForConditionalGeneration
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processor = WhisperProcessor.from_pretrained("openai/whisper-large")
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model = SpeechT5ForSpeechToText.from_pretrained("openai/whisper-large")
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model.config.forced_decoder_ids = WhisperProcessor.get_decoder_prompt_ids(language="english", task="transcribe")
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def process_audio(sampling_rate, waveform):
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# convert from int16 to floating point
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waveform = waveform / 32678.0
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# convert to mono if stereo
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if len(waveform.shape) > 1:
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waveform = librosa.to_mono(waveform.T)
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# resample to 16 kHz if necessary
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if sampling_rate != 16000:
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waveform = librosa.resample(waveform, orig_sr=sampling_rate, target_sr=16000)
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# limit to 30 seconds
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waveform = waveform[:16000*30]
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# make PyTorch tensor
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waveform = torch.tensor(waveform)
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return waveform
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def predict(audio, mic_audio=None):
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# audio = tuple (sample_rate, frames) or (sample_rate, (frames, channels))
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if mic_audio is not None:
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sampling_rate, waveform = mic_audio
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elif audio is not None:
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sampling_rate, waveform = audio
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else:
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return "(please provide audio)"
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waveform = process_audio(sampling_rate, waveform)
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input_features = processor(waveform, sampling_rate=16000, return_tensors="pt").input_features
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predicted_ids = model.generate(input_features, max_length=400)
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transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)
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return transcription[0]
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title = "Demo for Whisper -> Something -> XLS-R"
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description = """
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<b>How to use:</b> Upload an audio file or record using the microphone. The audio is converted to mono and resampled to 16 kHz before
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being passed into the model. The output is the text transcription of the audio.
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"""
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gr.Interface(
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fn=predict,
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inputs=[
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gr.Audio(label="Upload Speech", source="upload", type="numpy"),
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gr.Audio(label="Record Speech", source="microphone", type="numpy"),
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],
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outputs=[
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gr.Text(label="Transcription"),
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],
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title=title,
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article=article,
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).launch()
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