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Update app.py
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app.py
CHANGED
@@ -5,11 +5,8 @@ import torch
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import spaces
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import numpy as np
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@spaces.GPU(duration=120)
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def transcribe_and_respond(audio_file, chat_history):
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try:
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pipe = transformers.pipeline(
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model='sarvamai/shuka_v1',
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@@ -21,54 +18,35 @@ def transcribe_and_respond(audio_file, chat_history):
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# Load the audio file
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audio, sr = librosa.load(audio_file, sr=16000)
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#
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print(f"Audio dtype: {audio.dtype}, Audio shape: {audio.shape}, Sample rate: {sr}")
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# Debug: Print the
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print(f"
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# Call the model with the
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output = pipe({'audio': audio, 'turns': turns, 'sampling_rate': sr}, max_new_tokens=512)
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#
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turns.append({'role': 'system', 'content': output})
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# Debug: Print the model's response
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print(f"Model output: {output}")
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chat_history_for_display = []
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for turn in turns:
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if turn['role'] == 'user':
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chat_history_for_display.append(("User", "🗣️ (Spoken Audio)"))
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else:
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chat_history_for_display.append(("AI", turn['content']))
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return chat_history_for_display, turns # Return the formatted chat history for display and the updated history
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except Exception as e:
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return f"Error: {str(e)}"
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# Define the Gradio interface
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iface = gr.Interface(
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fn=transcribe_and_respond,
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inputs=
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gr.Chatbot(label="Conversation History"), # Display the conversation
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gr.State([]) # Hidden state to keep track of the updated conversation history
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],
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title="Shuka demo",
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description="shuka live demo",
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live=True, # Enable live mode for real-time interaction
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allow_flagging="auto",
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# enable_queue=True
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)
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if __name__ == "__main__":
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import spaces
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import numpy as np
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@spaces.GPU(duration=20)
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def transcribe_and_respond(audio_file):
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try:
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pipe = transformers.pipeline(
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model='sarvamai/shuka_v1',
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# Load the audio file
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audio, sr = librosa.load(audio_file, sr=16000)
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# Print audio properties for debugging
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print(f"Audio dtype: {audio.dtype}, Audio shape: {audio.shape}, Sample rate: {sr}")
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turns = [
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{'role': 'system', 'content': 'Respond naturally and informatively.'},
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{'role': 'user', 'content': '<|audio|>'}
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]
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# Debug: Print the initial turns
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print(f"Initial turns: {turns}")
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# Call the model with the audio and prompt
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output = pipe({'audio': audio, 'turns': turns, 'sampling_rate': sr}, max_new_tokens=512)
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# Debug: Print the final output from the model
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print(f"Model output: {output}")
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return output
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except Exception as e:
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return f"Error: {str(e)}"
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iface = gr.Interface(
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fn=transcribe_and_respond,
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inputs=gr.Audio(sources="microphone", type="filepath"),
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outputs="text",
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title="Live Transcription and Response",
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description="Speak into your microphone, and the model will respond naturally and informatively.",
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live=True
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)
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if __name__ == "__main__":
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