Update app.py
Browse files
app.py
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import gradio as gr
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from transformers import pipeline, AutoModelForCausalLM, AutoTokenizer
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from rapidfuzz import process, fuzz
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import soundfile as sf
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import numpy as np
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import torch
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#
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# 1
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#
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asr = pipeline(
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"automatic-speech-recognition",
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model="vhdm/whisper-large-fa-v1",
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device=-1
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)
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#
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# 2
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#
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llm_model_id = "tiiuae/falcon-rw-1b"
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tokenizer = AutoTokenizer.from_pretrained(llm_model_id)
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llm_model = AutoModelForCausalLM.from_pretrained(
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llm_model_id,
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torch_dtype=torch.float32
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).to("cpu")
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def ask_llm(prompt
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inputs = tokenizer(prompt, return_tensors="pt").to(llm_model.device)
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with torch.no_grad():
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outputs = llm_model.generate(**inputs, max_new_tokens=max_new_tokens)
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return tokenizer.decode(outputs[0], skip_special_tokens=True)
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#
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# 3
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#
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def text_to_speech_save(text: str, out_path: str = "response.wav") -> str:
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"""
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Use the text-to-speech pipeline to synthesize `text` and save to `out_path`.
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Returns the path on success or raises exception on failure.
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"""
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# pipeline may return a dict or list depending on versions; handle both
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result = tts_pipeline(text)
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if isinstance(result, list):
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entry = result[0]
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else:
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entry = result
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audio = entry.get("audio") if isinstance(entry, dict) else None
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sr = entry.get("sampling_rate", 16000) if isinstance(entry, dict) else 16000
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if audio is None:
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# some pipeline versions return numpy array directly
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audio = result if isinstance(result, np.ndarray) else None
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return out_path
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#
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# 4
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#
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"ΩΨ±Ψ―": ["ΩΨ±Ψ―", "ΩΩΨ±Ψ―", "ΩΩΨ±Ψ―"],
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"Ϊ©Ψ§Ω
ΩΎΫΩΨͺΨ±": ["Ϊ©Ψ§Ω
ΩΎΫΩΨͺΨ±", "Ϊ©Ψ§Ω
ΩΎΫΩΨͺΨ±Ω"],
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"ΩΩΨ΄ Ω
Ψ΅ΩΩΨΉΫ": ["ΩΩΨ΄ Ω
Ψ΅ΩΩΨΉΫ", "ΩΩΨ΄ Ψ΅ΩΨΉΨͺΫ"],
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"Ω
Ψ§Ψ΄ΫΩ": ["Ω
Ψ§Ψ΄ΫΩ", "Ω
Ψ§Ψ΄ΫΩΩ"],
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}
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def replace_fuzzy(text: str, vocab_map: dict, threshold: int = 85) -> str:
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"""
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Replace near-matches in `text` with canonical targets from vocab_map.
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Handles rapidfuzz.extractOne return types (object or tuple).
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"""
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if not text:
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return text
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for target, alternatives in vocab_map.items():
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try:
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res = process.extractOne(text, alternatives, scorer=fuzz.partial_ratio)
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except Exception:
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res = None
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if not res:
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continue
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# res may be an Extracted object or tuple
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if hasattr(res, "value") and hasattr(res, "score"):
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match = res.value
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score = res.score
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else:
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# tuple like (match, score, idx) or (match, score)
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match = res[0]
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score = res[1] if len(res) > 1 else 0
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if score >= threshold:
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# replace only the first occurrence to avoid accidental global replacement
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text = text.replace(match, target, 1)
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return text
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# ----------------------------
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# 5) Full pipeline function
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# ----------------------------
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def full_pipeline(audio_file: str):
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"""
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audio_file is a filepath (Gradio with type='filepath' sends a path for mic/upload).
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Returns (text_output_str, path_to_tts_wav or None).
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"""
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if not audio_file:
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return "No audio input detected.", None
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# 1) ASR
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try:
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except Exception as e:
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return f"ASR error: {e}", None
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if raw_text is None:
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raw_text = ""
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# 2) fuzzy replacement
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corrected_text = replace_fuzzy(raw_text, custom_vocab_map, threshold=85)
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# 3) LLM reply
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try:
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except Exception as e:
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# 4) TTS (synthesize LLM reply)
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try:
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except Exception as e:
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return f"
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return convo, audio_out
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#
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#
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#
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iface = gr.Interface(
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fn=full_pipeline,
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inputs=gr.Audio(type="filepath", label="Record or upload audio"),
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outputs=[gr.Textbox(label="Conversation"), gr.Audio(label="TTS Response")],
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title="Persian Voice Assistant",
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description="ASR β LLM β TTS
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)
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if __name__ == "__main__":
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import gradio as gr
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from transformers import pipeline, AutoModelForCausalLM, AutoTokenizer, SpeechT5Processor, SpeechT5ForTextToSpeech
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import torch
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import soundfile as sf
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# --------------------------
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# 1. ASR (speech to text)
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# --------------------------
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asr = pipeline(
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task="automatic-speech-recognition",
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model="vhdm/whisper-large-fa-v1",
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device=-1
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)
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# --------------------------
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# 2. Language Model (LLM)
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# --------------------------
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llm_model_id = "tiiuae/falcon-rw-1b"
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tokenizer = AutoTokenizer.from_pretrained(llm_model_id)
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llm_model = AutoModelForCausalLM.from_pretrained(
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llm_model_id,
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torch_dtype=torch.float32
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).to("cpu")
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def ask_llm(prompt, max_new_tokens=200):
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inputs = tokenizer(prompt, return_tensors="pt").to(llm_model.device)
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with torch.no_grad():
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outputs = llm_model.generate(**inputs, max_new_tokens=max_new_tokens)
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return tokenizer.decode(outputs[0], skip_special_tokens=True)
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# --------------------------
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# 3. TTS (text-to-speech) using SpeechT5
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# --------------------------
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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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# Random speaker embedding (can be replaced with a fixed one for consistency)
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speaker_embedding = torch.randn(1, 512)
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def text_to_speech(text, out_path="output.wav"):
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inputs = processor(text=text, return_tensors="pt")
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with torch.no_grad():
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speech = tts_model.generate_speech(inputs["input_ids"], speaker_embedding)
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sf.write(out_path, speech.numpy(), 16000)
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return out_path
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# --------------------------
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# 4. Full pipeline function
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# --------------------------
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def full_pipeline(audio_file):
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if not audio_file:
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return "No audio input detected.", None
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try:
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result = asr(audio_file, chunk_length_s=30, stride_length_s=[5, 5])
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except Exception as e:
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return f"ASR error: {e}", None
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user_text = result.get("text", "")
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try:
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llm_response = ask_llm(user_text)
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except Exception as e:
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return f"Assistant generation error: {e}", None
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try:
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audio_path = text_to_speech(llm_response, "response.wav")
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except Exception as e:
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return f"TTS error: {e}", None
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return f"User said: {user_text}\nAssistant: {llm_response}", audio_path
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# --------------------------
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# 5. Gradio Interface
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# --------------------------
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iface = gr.Interface(
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fn=full_pipeline,
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inputs=gr.Audio(type="filepath", label="Record or upload audio"),
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outputs=[gr.Textbox(label="Conversation"), gr.Audio(label="TTS Response")],
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title="Persian Voice Assistant",
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description="ASR β LLM β TTS"
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)
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if __name__ == "__main__":
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