app.py
Browse filesusing tiny
app.py
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import torch
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import gradio as gr
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import time
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import numpy as np
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import scipy.io.wavfile
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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# β
1οΈβ£ Force Model to Run on CPU
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device = "cpu"
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torch_dtype = torch.float32 # Use CPU-friendly float type
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MODEL_NAME = "openai/whisper-tiny" # β
Switched to smallest model for fastest performance
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# β
2οΈβ£ Load Whisper Tiny Model on CPU
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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MODEL_NAME, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
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)
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model.to(device)
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# β
3οΈβ£ Load Processor & Pipeline
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processor = AutoProcessor.from_pretrained(MODEL_NAME)
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pipe = pipeline(
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task="automatic-speech-recognition",
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model=model,
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tokenizer=processor.tokenizer,
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feature_extractor=processor.feature_extractor,
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chunk_length_s=2, # β
Process in 2-second chunks for ultra-low latency
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torch_dtype=torch_dtype,
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device=device,
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)
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# β
4οΈβ£ Real-Time Streaming Transcription (Microphone)
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def stream_transcribe(stream, new_chunk):
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start_time = time.time()
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try:
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sr, y = new_chunk
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# β
Convert stereo to mono
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if y.ndim > 1:
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y = y.mean(axis=1)
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y = y.astype(np.float32)
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y /= np.max(np.abs(y))
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# β
Append to Stream
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if stream is not None:
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stream = np.concatenate([stream, y])
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else:
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stream = y
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# β
Run Transcription
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transcription = pipe({"sampling_rate": sr, "raw": stream})["text"]
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latency = time.time() - start_time
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return stream, transcription, f"{latency:.2f} sec"
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except Exception as e:
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print(f"Error: {e}")
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return stream, str(e), "Error"
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# β
5οΈβ£ Transcription for File Upload
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def transcribe(inputs, previous_transcription):
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start_time = time.time()
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try:
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# β
Convert file input to correct format
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sample_rate, audio_data = inputs
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transcription = pipe({"sampling_rate": sample_rate, "raw": audio_data})["text"]
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previous_transcription += transcription
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latency = time.time() - start_time
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return previous_transcription, f"{latency:.2f} sec"
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except Exception as e:
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print(f"Error: {e}")
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return previous_transcription, "Error"
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# β
6οΈβ£ Clear Function
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def clear():
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return ""
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# β
7οΈβ£ Gradio Interface (Microphone Streaming)
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with gr.Blocks() as microphone:
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gr.Markdown(f"# Whisper Tiny - Real-Time Transcription (CPU) ποΈ")
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gr.Markdown(f"Using [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) for ultra-fast speech-to-text.")
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with gr.Row():
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input_audio_microphone = gr.Audio(sources=["microphone"], type="numpy", streaming=True)
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output = gr.Textbox(label="Live Transcription", value="")
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latency_textbox = gr.Textbox(label="Latency (seconds)", value="0.0")
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with gr.Row():
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clear_button = gr.Button("Clear Output")
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state = gr.State()
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input_audio_microphone.stream(
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stream_transcribe, [state, input_audio_microphone],
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[state, output, latency_textbox], time_limit=30, stream_every=1
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)
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clear_button.click(clear, outputs=[output])
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# β
8οΈβ£ Gradio Interface (File Upload)
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with gr.Blocks() as file:
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gr.Markdown(f"# Upload Audio File for Transcription π΅")
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gr.Markdown(f"Using [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) for speech-to-text.")
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with gr.Row():
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input_audio = gr.Audio(sources=["upload"], type="numpy")
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output = gr.Textbox(label="Transcription", value="")
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latency_textbox = gr.Textbox(label="Latency (seconds)", value="0.0")
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with gr.Row():
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submit_button = gr.Button("Submit")
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clear_button = gr.Button("Clear Output")
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submit_button.click(transcribe, [input_audio, output], [output, latency_textbox])
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clear_button.click(clear, outputs=[output])
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# β
9οΈβ£ Final Gradio App (Supports Microphone & File Upload)
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with gr.Blocks(theme=gr.themes.Ocean()) as demo:
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gr.TabbedInterface([microphone, file], ["Microphone", "Upload Audio"])
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# β
1οΈβ£0οΈβ£ Run Gradio Locally
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if __name__ == "__main__":
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demo.launch()
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