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app.py
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import torch
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from transformers import AutoModelForCausalLM, AutoTokenizer
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from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan
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from datasets import load_dataset
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from datasets import load_dataset
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from transformers import pipeline
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import numpy as np
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import gradio as gr
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asr_model = pipeline(task = "automatic-speech-recognition",
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model = "distil-whisper/distil-small.en")
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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llm_model = AutoModelForCausalLM.from_pretrained("OuteAI/Lite-Mistral-150M-v2-Instruct").to(device)
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tokenizer = AutoTokenizer.from_pretrained("OuteAI/Lite-Mistral-150M-v2-Instruct")
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def generate_response(message: str, temperature: float = 0.3, repetition_penalty: float = 1.2) -> str:
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# Apply the chat template and convert to PyTorch tensors
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messages = [
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{"role": "system",
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"content": "You are my best close friend. you are full of human emotions and you have feelings. provide short responses."},
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{"role": "user", "content": message}
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]
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input_ids = tokenizer.apply_chat_template(
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messages, add_generation_prompt=True, return_tensors="pt"
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).to(device)
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# Generate the response
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output = llm_model.generate(
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input_ids,
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max_length = 150,
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pad_token_id=tokenizer.eos_token_id,
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temperature=temperature,
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repetition_penalty=repetition_penalty,
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do_sample=True
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)
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# Decode the generated output
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generated_text = tokenizer.decode(output[0], skip_special_tokens=True)
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return generated_text.split('\n assistant\n')[1]
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processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts")
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tts_model = SpeechT5ForTextToSpeech.from_pretrained("microsoft/speecht5_tts")
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vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")
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embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation")
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speaker_embeddings = torch.tensor(embeddings_dataset[7300]["xvector"]).unsqueeze(0)
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def text_to_speech(input_text):
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inputs = processor(text=input_text, return_tensors="pt")
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speech = tts_model.generate_speech(inputs["input_ids"], speaker_embeddings, vocoder=vocoder)
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return speech
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def spoken_llm(input_voice_file):
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if input_voice_file is None:
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gr.Warning("No input audio")
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return None
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asr_text = asr_model(input_voice_file)['text']
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print("\n\nASR\n\n")
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print(asr_text)
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llm_response = generate_response(asr_text)
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print("\n\nLLM\n\n")
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print(llm_response)
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audio_out = text_to_speech(llm_response)
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print("\n\nTTS\n\n")
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rate = 17000
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return rate, (audio_out.cpu().numpy().reshape(-1)*2e4).astype(np.int16)
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interface = gr.Interface(fn = spoken_llm, inputs = gr.Audio(sources = "microphone", type = "filepath"), outputs = "audio")
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interface.launch()
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