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Update app.py
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app.py
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@@ -3,52 +3,40 @@ import torch
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import torchaudio
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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# Load the
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model_id = "facebook/wav2vec2-base-960h"
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processor = Wav2Vec2Processor.from_pretrained(model_id)
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model = Wav2Vec2ForCTC.from_pretrained(model_id)
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# Transcription function with optimization
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def transcribe(audio_file, progress=gr.Progress()):
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if audio_file is None:
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return "⚠️ No audio received."
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print(f"📥 Received file path: {audio_file}") # ✅ Log for debugging
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progress(0, desc="Loading audio...")
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waveform, sample_rate = torchaudio.load(audio_file)
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if sample_rate != 16000:
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progress(0.3, desc="Resampling audio...")
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waveform = torchaudio.functional.resample(waveform, orig_freq=sample_rate, new_freq=16000)
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sample_rate = 16000
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if waveform.shape[0] > 1:
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progress(0.5, desc="Converting to mono...")
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waveform = waveform.mean(dim=0).unsqueeze(0)
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progress(0.7, desc="Processing audio...")
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input_values = processor(waveform.squeeze().numpy(), sampling_rate=sample_rate, return_tensors="pt").input_values
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with torch.no_grad():
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progress(0.8, desc="Transcribing...")
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(predicted_ids)[0]
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progress(1.0, desc="Done!")
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return transcription.lower()
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# Gradio UI with POST API support
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demo = gr.Interface(
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fn=transcribe,
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inputs=gr.Audio(sources=["microphone"], type="filepath", label="🎤 Speak now"),
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outputs=gr.Textbox(label="📝 Transcription"),
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title="Wav2Vec2 Speech Transcription",
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description="Speak into the microphone and get a transcription using Wav2Vec2-base
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)
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demo.launch(api=True)
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import torchaudio
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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# Load the model
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model_id = "facebook/wav2vec2-base-960h"
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processor = Wav2Vec2Processor.from_pretrained(model_id)
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model = Wav2Vec2ForCTC.from_pretrained(model_id)
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def transcribe(audio_file, progress=gr.Progress()):
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if audio_file is None:
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return "⚠️ No audio received."
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waveform, sample_rate = torchaudio.load(audio_file)
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if sample_rate != 16000:
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waveform = torchaudio.functional.resample(waveform, orig_freq=sample_rate, new_freq=16000)
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sample_rate = 16000
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if waveform.shape[0] > 1:
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waveform = waveform.mean(dim=0).unsqueeze(0)
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input_values = processor(waveform.squeeze().numpy(), sampling_rate=sample_rate, return_tensors="pt").input_values
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with torch.no_grad():
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(predicted_ids)[0]
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return transcription.lower()
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demo = gr.Interface(
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fn=transcribe,
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inputs=gr.Audio(sources=["microphone"], type="filepath", label="🎤 Speak now"),
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outputs=gr.Textbox(label="📝 Transcription"),
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title="Wav2Vec2 Speech Transcription",
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description="Speak into the microphone and get a transcription using Wav2Vec2-base.",
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flagging_mode="never"
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)
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demo.launch()
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