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| 1 |
+
import gradio as gr
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| 2 |
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import torch
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| 3 |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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| 4 |
+
import time
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| 5 |
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import os
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| 6 |
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import numpy as np
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| 7 |
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import soundfile as sf
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| 8 |
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import librosa
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| 9 |
+
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| 10 |
+
# --- Configuration ---
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| 11 |
+
# Device selection (GPU if available, else CPU)
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| 12 |
+
device = "cuda:0" if torch.cuda.is_available() else "cpu"
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| 13 |
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
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| 14 |
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print(f"Using device: {device}")
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| 15 |
+
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| 16 |
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# STT Model (Use smaller model for lower latency)
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| 17 |
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stt_model_id = "openai/whisper-tiny" # Or "openai/whisper-base". Avoid larger models for streaming.
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| 18 |
+
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| 19 |
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# Summarization Model
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| 20 |
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summarizer_model_id = "sshleifer/distilbart-cnn-6-6" # Use a distilled/smaller model for speed
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| 21 |
+
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| 22 |
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# Summarization Interval (seconds) - How often to regenerate the summary
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| 23 |
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SUMMARY_INTERVAL = 30.0 # Summarize every 30 seconds
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| 24 |
+
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| 25 |
+
# --- Load Models ---
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| 26 |
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# (Keep the model loading code exactly the same as before)
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| 27 |
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print("Loading STT model...")
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| 28 |
+
stt_model = AutoModelForSpeechSeq2Seq.from_pretrained(
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| 29 |
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stt_model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
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| 30 |
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)
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| 31 |
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stt_model.to(device)
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| 32 |
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processor = AutoProcessor.from_pretrained(stt_model_id)
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| 33 |
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stt_pipeline = pipeline(
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| 34 |
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"automatic-speech-recognition",
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| 35 |
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model=stt_model,
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| 36 |
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tokenizer=processor.tokenizer,
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| 37 |
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feature_extractor=processor.feature_extractor,
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| 38 |
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max_new_tokens=128,
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| 39 |
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chunk_length_s=30,
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| 40 |
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batch_size=16,
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| 41 |
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torch_dtype=torch_dtype,
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| 42 |
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device=device,
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| 43 |
+
)
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| 44 |
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print("STT model loaded.")
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| 45 |
+
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| 46 |
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print("Loading Summarization pipeline...")
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| 47 |
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summarizer = pipeline(
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| 48 |
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"summarization",
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| 49 |
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model=summarizer_model_id,
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| 50 |
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device=device
|
| 51 |
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)
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| 52 |
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print("Summarization pipeline loaded.")
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| 53 |
+
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| 54 |
+
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| 55 |
+
# --- Helper Functions ---
|
| 56 |
+
# (Keep the format_summary_as_bullets function exactly the same)
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| 57 |
+
def format_summary_as_bullets(summary_text):
|
| 58 |
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"""Attempts to format a summary text block into bullet points."""
|
| 59 |
+
if not summary_text:
|
| 60 |
+
return ""
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| 61 |
+
# Simple approach: split by sentences and add bullets.
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| 62 |
+
# More advanced NLP could be used here.
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| 63 |
+
sentences = summary_text.replace(". ", ".\n- ").split('\n')
|
| 64 |
+
bullet_summary = "- " + "\n".join(sentences).strip()
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| 65 |
+
# Remove potential empty bullets
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| 66 |
+
bullet_summary = "\n".join([line for line in bullet_summary.split('\n') if line.strip() not in ['-', '']])
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| 67 |
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return bullet_summary
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| 68 |
+
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| 69 |
+
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| 70 |
+
# --- Processing Function for Streaming ---
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| 71 |
+
# (Keep the process_audio_stream function exactly the same)
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| 72 |
+
# This function ONLY processes audio, it doesn't interact with the webcam video
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| 73 |
+
def process_audio_stream(
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| 74 |
+
new_chunk_tuple, # Gradio streaming yields (sample_rate, numpy_data)
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| 75 |
+
accumulated_transcript_state, # gr.State holding the full text
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| 76 |
+
last_summary_time_state, # gr.State holding the timestamp of the last summary
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| 77 |
+
current_summary_state # gr.State holding the last generated summary
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| 78 |
+
):
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| 79 |
+
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| 80 |
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if new_chunk_tuple is None:
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| 81 |
+
# Initial call or stream ended, return current state
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| 82 |
+
return accumulated_transcript_state, current_summary_state, accumulated_transcript_state, last_summary_time_state, current_summary_state
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| 83 |
+
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| 84 |
+
sample_rate, audio_chunk = new_chunk_tuple
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| 85 |
+
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| 86 |
+
if audio_chunk is None or sample_rate is None or audio_chunk.size == 0:
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| 87 |
+
# Handle potential empty chunks gracefully
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| 88 |
+
return accumulated_transcript_state, current_summary_state, accumulated_transcript_state, last_summary_time_state, current_summary_state
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| 89 |
+
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| 90 |
+
print(f"Received chunk: {audio_chunk.shape}, Sample Rate: {sample_rate}, Duration: {len(audio_chunk)/sample_rate:.2f}s")
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| 91 |
+
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| 92 |
+
# Ensure audio is float32 and mono, as Whisper expects
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| 93 |
+
if audio_chunk.dtype != np.float32:
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| 94 |
+
# Normalize assuming input is int16
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| 95 |
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# Adjust if your microphone provides different integer types
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| 96 |
+
audio_chunk = audio_chunk.astype(np.float32) / 32768.0 # Max value for int16 is 32767
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| 97 |
+
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| 98 |
+
# --- 1. Transcribe the new chunk ---
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| 99 |
+
new_text = ""
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| 100 |
+
try:
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| 101 |
+
result = stt_pipeline({"sampling_rate": sample_rate, "raw": audio_chunk.copy()})
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| 102 |
+
new_text = result["text"].strip() if result["text"] else ""
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| 103 |
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print(f"Transcription chunk: '{new_text}'")
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| 104 |
+
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| 105 |
+
except Exception as e:
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| 106 |
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print(f"Error during transcription chunk: {e}")
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| 107 |
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new_text = f"[Transcription Error: {e}]"
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| 108 |
+
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| 109 |
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# --- 2. Update Accumulated Transcript ---
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| 110 |
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if accumulated_transcript_state and not accumulated_transcript_state.endswith((" ", "\n")) and new_text:
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| 111 |
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updated_transcript = accumulated_transcript_state + " " + new_text
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| 112 |
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else:
|
| 113 |
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updated_transcript = accumulated_transcript_state + new_text
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| 114 |
+
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| 115 |
+
# --- 3. Periodic Summarization ---
|
| 116 |
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current_time = time.time()
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| 117 |
+
new_summary = current_summary_state # Keep the old summary by default
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| 118 |
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updated_last_summary_time = last_summary_time_state
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| 119 |
+
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| 120 |
+
# Check transcript length to avoid summarizing tiny bits of text too early
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| 121 |
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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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| 122 |
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print(f"Summarizing transcript (length: {len(updated_transcript)})...")
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| 123 |
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try:
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| 124 |
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# Summarize the *entire* transcript up to this point
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| 125 |
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summary_result = summarizer(updated_transcript, max_length=150, min_length=30, do_sample=False)
|
| 126 |
+
if summary_result and isinstance(summary_result, list):
|
| 127 |
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raw_summary = summary_result[0]['summary_text']
|
| 128 |
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new_summary = format_summary_as_bullets(raw_summary)
|
| 129 |
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updated_last_summary_time = current_time # Update time only on successful summary
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| 130 |
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print("Summary updated.")
|
| 131 |
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else:
|
| 132 |
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print("Summarization did not produce expected output.")
|
| 133 |
+
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| 134 |
+
except Exception as e:
|
| 135 |
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print(f"Error during summarization: {e}")
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| 136 |
+
# Display error in summary box but keep the last known good summary in state
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| 137 |
+
# To avoid overwriting a potentially useful summary with just an error message
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| 138 |
+
# We return the error message for display, but not update summary_state with it
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| 139 |
+
error_display_summary = f"[Summarization Error]\n\nLast good summary:\n{current_summary_state}"
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| 140 |
+
return updated_transcript, error_display_summary, updated_transcript, last_summary_time_state, current_summary_state
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| 141 |
+
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| 142 |
+
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| 143 |
+
# --- 4. Return Updated State and Outputs ---
|
| 144 |
+
return updated_transcript, new_summary, updated_transcript, updated_last_summary_time, new_summary
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| 145 |
+
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| 146 |
+
|
| 147 |
+
# --- Gradio Interface ---
|
| 148 |
+
print("Creating Gradio interface...")
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| 149 |
+
with gr.Blocks() as demo:
|
| 150 |
+
gr.Markdown("# Real-Time Meeting Notes with Webcam View")
|
| 151 |
+
gr.Markdown("Speak into your microphone. Transcription appears below. Summary updates periodically.")
|
| 152 |
+
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| 153 |
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# State variables to store data between stream calls
|
| 154 |
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transcript_state = gr.State("") # Holds the full transcript
|
| 155 |
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last_summary_time = gr.State(0.0) # Holds the time the summary was last generated
|
| 156 |
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summary_state = gr.State("") # Holds the current bullet point summary
|
| 157 |
+
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| 158 |
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with gr.Row():
|
| 159 |
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with gr.Column(scale=1):
|
| 160 |
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# Input: Microphone stream
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| 161 |
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audio_stream = gr.Audio(sources=["microphone"], streaming=True, label="Live Microphone Input", type="numpy")
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| 162 |
+
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| 163 |
+
# NEW: Webcam Display
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| 164 |
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# Use gr.Image which is simpler for just displaying webcam feed
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| 165 |
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# live=True makes it update continuously
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| 166 |
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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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| 167 |
+
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| 168 |
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with gr.Column(scale=2):
|
| 169 |
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transcription_output = gr.Textbox(label="Full Transcription", lines=15, interactive=False) # Display only
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| 170 |
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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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| 171 |
+
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| 172 |
+
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| 173 |
+
# Connect the streaming audio input to the processing function
|
| 174 |
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# Note: The webcam component runs independently in the browser, it doesn't feed data here
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| 175 |
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audio_stream.stream(
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| 176 |
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fn=process_audio_stream,
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| 177 |
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inputs=[audio_stream, transcript_state, last_summary_time, summary_state],
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| 178 |
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outputs=[transcription_output, summary_output, transcript_state, last_summary_time, summary_state],
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| 179 |
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)
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| 180 |
+
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| 181 |
+
# Add a button to clear the state if needed
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| 182 |
+
def clear_state_values():
|
| 183 |
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print("Clearing state.")
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| 184 |
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return "", "", 0.0, "" # Clear transcript display, summary display, reset time state, clear summary state
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| 185 |
+
# Need separate function to clear states vs displays if they differ
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| 186 |
+
def clear_state():
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| 187 |
+
return "", 0.0, "" # Clear transcript_state, last_summary_time, summary_state
|
| 188 |
+
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| 189 |
+
clear_button = gr.Button("Clear Transcript & Summary")
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| 190 |
+
# This button clears the display textboxes AND resets the internal states
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| 191 |
+
clear_button.click(
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| 192 |
+
fn=lambda: ("", "", "", 0.0, ""), # Return empty values for all outputs/states
|
| 193 |
+
inputs=[],
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| 194 |
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outputs=[transcription_output, summary_output, transcript_state, last_summary_time, summary_state]
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| 195 |
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)
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| 196 |
+
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| 197 |
+
|
| 198 |
+
print("Launching Gradio interface...")
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| 199 |
+
demo.queue() # Enable queue for handling multiple requests/stream chunks
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| 200 |
+
demo.launch(debug=True, share=True) # share=True for Colab public link
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