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Update app.py
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app.py
CHANGED
@@ -9,9 +9,43 @@ device = "cuda" if torch.cuda.is_available() else "cpu"
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# Load processor & model
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model_name = "cdactvm/w2v-bert-punjabi" # Change if using a Punjabi ASR model
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processor = Wav2Vec2BertProcessor.from_pretrained(model_name)
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# Load audio file
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waveform, sample_rate = torchaudio.load(audio_path)
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@@ -29,53 +63,37 @@ def transcribe(audio_path):
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# Get logits & transcribe
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with torch.no_grad():
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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
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if __name__ == "__main__":
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app.launch()
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# import gradio as gr
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# import torch
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# from transformers import pipeline
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# # Set device
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# device = "cuda" if torch.cuda.is_available() else "cpu"
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# # Load ASR pipeline
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# asr_pipeline = pipeline(
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# "automatic-speech-recognition",
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# model="cdactvm/w2v-bert-punjabi", # Replace with a Punjabi ASR model if available
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# torch_dtype=torch.bfloat16,
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# device=0 if torch.cuda.is_available() else -1 # GPU (0) or CPU (-1)
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# )
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# def transcribe(audio_path):
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# # Run inference
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# result = asr_pipeline(audio_path)
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# return result["text"]
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# # Gradio Interface
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# app = gr.Interface(
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# fn=transcribe,
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# inputs=gr.Audio(sources="upload", type="filepath"),
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# outputs="text",
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# title="Punjabi Speech-to-Text",
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# description="Upload an audio file and get the transcription in Punjabi."
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# )
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# if __name__ == "__main__":
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# app.launch()
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# Load processor & model
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model_name = "cdactvm/w2v-bert-punjabi" # Change if using a Punjabi ASR model
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processor = Wav2Vec2BertProcessor.from_pretrained(model_name)
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# Loading the original model.
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original_model=Wav2Vec2BertForCTC.from_pretrained(model_name)
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# Explicitly allow Wav2Vec2BertForCTC during unpickling3+
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torch.serialization.add_safe_globals([Wav2Vec2BertForCTC])
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# Load the full quantized model
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quantized_model = torch.load("model_name", weights_only=False)
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quantized_model.eval()
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#####################################################
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# recognize speech using original model
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def transcribe_original_model(audio_path):
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# Load audio file
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waveform, sample_rate = torchaudio.load(audio_path)
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# Convert stereo to mono (if needed)
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if waveform.shape[0] > 1:
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waveform = torch.mean(waveform, dim=0, keepdim=True)
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# Resample to 16kHz
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if sample_rate != 16000:
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waveform = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(waveform)
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# Process audio
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inputs = processor(waveform.squeeze(0), sampling_rate=16000, return_tensors="pt")
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inputs = {key: val.to(device, dtype=torch.bfloat16) for key, val in inputs.items()}
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# Get logits & transcribe
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with torch.no_grad():
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logits = original_model(**inputs).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
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# recognize speech using quantized model.
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def transcribe_quantized_model(audio_path):
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# Load audio file
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waveform, sample_rate = torchaudio.load(audio_path)
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# Get logits & transcribe
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with torch.no_grad():
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logits = quantized_model(**inputs).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
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def select_lng(lng, mic=None, file=None):
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if mic is not None:
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audio = mic
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elif file is not None:
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audio = file
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else:
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return "You must either provide a mic recording or a file"
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if lng == "original_model":
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return transcribe_original_model(audio)
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elif lng == "quantized_model":
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return transcribe_quantized_model(audio)
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# Gradio Interface
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demo=gr.Interface(
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fn=select_lng,
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inputs=[
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gr.Dropdown(["original_model","quantized_model"],label="Select Model"),
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gr.Audio(sources=["microphone","upload"], type="filepath"),
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],
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outputs=["textbox"],
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title="Automatic Speech Recognition",
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description = "Upload an audio file and get the transcription in Punjabi.",
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)
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if __name__ == "__main__":
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app.launch()
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