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import gradio as gr |
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import torch |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline |
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import time |
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import os |
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import numpy as np |
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import soundfile as sf |
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import librosa |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 |
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print(f"Using device: {device}") |
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stt_model_id = "openai/whisper-tiny" # Or "openai/whisper-base". Avoid larger models for streaming. |
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summarizer_model_id = "sshleifer/distilbart-cnn-6-6" # Use a distilled/smaller model for speed |
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SUMMARY_INTERVAL = 30.0 # Summarize every 30 seconds |
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print("Loading STT model...") |
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stt_model = AutoModelForSpeechSeq2Seq.from_pretrained( |
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stt_model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True |
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) |
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stt_model.to(device) |
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processor = AutoProcessor.from_pretrained(stt_model_id) |
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stt_pipeline = pipeline( |
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"automatic-speech-recognition", |
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model=stt_model, |
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tokenizer=processor.tokenizer, |
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feature_extractor=processor.feature_extractor, |
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max_new_tokens=128, |
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chunk_length_s=30, |
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batch_size=16, |
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torch_dtype=torch_dtype, |
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device=device, |
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) |
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print("STT model loaded.") |
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print("Loading Summarization pipeline...") |
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summarizer = pipeline( |
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"summarization", |
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model=summarizer_model_id, |
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device=device |
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) |
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print("Summarization pipeline loaded.") |
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def format_summary_as_bullets(summary_text): |
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"""Attempts to format a summary text block into bullet points.""" |
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if not summary_text: |
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return "" |
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# Simple approach: split by sentences and add bullets. |
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# More advanced NLP could be used here. |
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sentences = summary_text.replace(". ", ".\n- ").split('\n') |
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bullet_summary = "- " + "\n".join(sentences).strip() |
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# Remove potential empty bullets |
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bullet_summary = "\n".join([line for line in bullet_summary.split('\n') if line.strip() not in ['-', '']]) |
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return bullet_summary |
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def process_audio_stream( |
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new_chunk_tuple, # Gradio streaming yields (sample_rate, numpy_data) |
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accumulated_transcript_state, # gr.State holding the full text |
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last_summary_time_state, # gr.State holding the timestamp of the last summary |
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current_summary_state # gr.State holding the last generated summary |
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): |
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if new_chunk_tuple is None: |
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# Initial call or stream ended, return current state |
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return accumulated_transcript_state, current_summary_state, accumulated_transcript_state, last_summary_time_state, current_summary_state |
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sample_rate, audio_chunk = new_chunk_tuple |
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if audio_chunk is None or sample_rate is None or audio_chunk.size == 0: |
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# Handle potential empty chunks gracefully |
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return accumulated_transcript_state, current_summary_state, accumulated_transcript_state, last_summary_time_state, current_summary_state |
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print(f"Received chunk: {audio_chunk.shape}, Sample Rate: {sample_rate}, Duration: {len(audio_chunk)/sample_rate:.2f}s") |
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# Ensure audio is float32 and mono, as Whisper expects |
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if audio_chunk.dtype != np.float32: |
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# Normalize assuming input is int16 |
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# Adjust if your microphone provides different integer types |
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audio_chunk = audio_chunk.astype(np.float32) / 32768.0 # Max value for int16 is 32767 |
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# --- 1. Transcribe the new chunk --- |
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new_text = "" |
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try: |
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result = stt_pipeline({"sampling_rate": sample_rate, "raw": audio_chunk.copy()}) |
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new_text = result["text"].strip() if result["text"] else "" |
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print(f"Transcription chunk: '{new_text}'") |
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except Exception as e: |
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print(f"Error during transcription chunk: {e}") |
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new_text = f"[Transcription Error: {e}]" |
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# --- 2. Update Accumulated Transcript --- |
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if accumulated_transcript_state and not accumulated_transcript_state.endswith((" ", "\n")) and new_text: |
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updated_transcript = accumulated_transcript_state + " " + new_text |
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else: |
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updated_transcript = accumulated_transcript_state + new_text |
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# --- 3. Periodic Summarization --- |
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current_time = time.time() |
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new_summary = current_summary_state # Keep the old summary by default |
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updated_last_summary_time = last_summary_time_state |
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# Check transcript length to avoid summarizing tiny bits of text too early |
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if updated_transcript and len(updated_transcript) > 50 and (current_time - last_summary_time_state > SUMMARY_INTERVAL): |
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print(f"Summarizing transcript (length: {len(updated_transcript)})...") |
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try: |
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# Summarize the *entire* transcript up to this point |
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summary_result = summarizer(updated_transcript, max_length=150, min_length=30, do_sample=False) |
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if summary_result and isinstance(summary_result, list): |
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raw_summary = summary_result[0]['summary_text'] |
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new_summary = format_summary_as_bullets(raw_summary) |
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updated_last_summary_time = current_time # Update time only on successful summary |
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print("Summary updated.") |
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else: |
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print("Summarization did not produce expected output.") |
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except Exception as e: |
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print(f"Error during summarization: {e}") |
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# Display error in summary box but keep the last known good summary in state |
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# To avoid overwriting a potentially useful summary with just an error message |
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# We return the error message for display, but not update summary_state with it |
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error_display_summary = f"[Summarization Error]\n\nLast good summary:\n{current_summary_state}" |
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return updated_transcript, error_display_summary, updated_transcript, last_summary_time_state, current_summary_state |
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# --- 4. Return Updated State and Outputs --- |
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return updated_transcript, new_summary, updated_transcript, updated_last_summary_time, new_summary |
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print("Creating Gradio interface...") |
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with gr.Blocks() as demo: |
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gr.Markdown("# Real-Time Meeting Notes with Webcam View") |
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gr.Markdown("Speak into your microphone. Transcription appears below. Summary updates periodically.") |
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# State variables to store data between stream calls |
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transcript_state = gr.State("") # Holds the full transcript |
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last_summary_time = gr.State(0.0) # Holds the time the summary was last generated |
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summary_state = gr.State("") # Holds the current bullet point summary |
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with gr.Row(): |
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with gr.Column(scale=1): |
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# Input: Microphone stream |
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audio_stream = gr.Audio(sources=["microphone"], streaming=True, label="Live Microphone Input", type="numpy") |
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# NEW: Webcam Display |
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# Use gr.Image which is simpler for just displaying webcam feed |
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# live=True makes it update continuously |
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webcam_view = gr.Image(sources=["webcam"], label="Your Webcam", streaming=True) # Use streaming=True for live view |
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with gr.Column(scale=2): |
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transcription_output = gr.Textbox(label="Full Transcription", lines=15, interactive=False) # Display only |
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summary_output = gr.Textbox(label=f"Bullet Point Summary (Updates ~every {SUMMARY_INTERVAL}s)", lines=10, interactive=False) # Display only |
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# Connect the streaming audio input to the processing function |
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# Note: The webcam component runs independently in the browser, it doesn't feed data here |
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audio_stream.stream( |
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fn=process_audio_stream, |
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inputs=[audio_stream, transcript_state, last_summary_time, summary_state], |
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outputs=[transcription_output, summary_output, transcript_state, last_summary_time, summary_state], |
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) |
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# Add a button to clear the state if needed |
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def clear_state_values(): |
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print("Clearing state.") |
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return "", "", 0.0, "" # Clear transcript display, summary display, reset time state, clear summary state |
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# Need separate function to clear states vs displays if they differ |
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def clear_state(): |
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return "", 0.0, "" # Clear transcript_state, last_summary_time, summary_state |
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clear_button = gr.Button("Clear Transcript & Summary") |
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# This button clears the display textboxes AND resets the internal states |
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clear_button.click( |
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fn=lambda: ("", "", "", 0.0, ""), # Return empty values for all outputs/states |
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inputs=[], |
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outputs=[transcription_output, summary_output, transcript_state, last_summary_time, summary_state] |
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) |
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print("Launching Gradio interface...") |
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demo.queue() # Enable queue for handling multiple requests/stream chunks |
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demo.launch(debug=True, share=True) # share=True for Colab public link |