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Update app.py
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app.py
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@@ -1,19 +1,43 @@
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import numpy as np
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import gradio as gr
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from dotenv import load_dotenv
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from fastrtc import (
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ReplyOnPause,
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Stream,
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get_current_context,
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get_hf_turn_credentials,
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get_hf_turn_credentials_async,
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get_stt_model,
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get_tts_model,
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WebRTCError,
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)
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import gradio as gr
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from huggingface_hub import InferenceClient
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load_dotenv()
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if hf_token is None or hf_token == "":
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raise WebRTCError("HF Token is required")
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llm_client = InferenceClient(provider="
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context = get_current_context()
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if context.webrtc_id not in conversations:
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messages = conversations[context.webrtc_id]
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transcription = stt_model.stt(audio)
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messages.append({"role": "user", "content": transcription})
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output = llm_client.chat.completions.create( # type: ignore
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additional_outputs=[chatbot],
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additional_outputs_handler=lambda old, new: new,
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ui_args={"title": "Talk To OpenAI GPT-OSS 20B (Powered by FastRTC ⚡️)"},
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)
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stream.ui.launch()
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import gradio as gr
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import numpy as np
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import torch
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from dotenv import load_dotenv
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from fastrtc import (
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AdditionalOutputs,
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ReplyOnPause,
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Stream,
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WebRTCError,
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audio_to_float32,
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get_current_context,
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get_hf_turn_credentials,
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get_hf_turn_credentials_async,
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get_stt_model,
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get_tts_model,
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)
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from huggingface_hub import InferenceClient
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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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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model_id = "openai/whisper-large-v3-turbo"
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
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)
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model.to(device)
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processor = AutoProcessor.from_pretrained(model_id)
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pipe = pipeline(
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"automatic-speech-recognition",
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model=model,
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tokenizer=processor.tokenizer,
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feature_extractor=processor.feature_extractor,
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torch_dtype=torch_dtype,
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device=device,
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)
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load_dotenv()
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if hf_token is None or hf_token == "":
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raise WebRTCError("HF Token is required")
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llm_client = InferenceClient(provider="auto", token=hf_token)
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result = pipe(
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{"array": audio_to_float32(audio[1]).squeeze(), "sampling_rate": audio[0]}
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)
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transcription = result["text"]
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context = get_current_context()
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if context.webrtc_id not in conversations:
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messages = conversations[context.webrtc_id]
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messages.append({"role": "user", "content": transcription})
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output = llm_client.chat.completions.create( # type: ignore
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additional_outputs=[chatbot],
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additional_outputs_handler=lambda old, new: new,
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ui_args={"title": "Talk To OpenAI GPT-OSS 20B (Powered by FastRTC ⚡️)"},
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time_limit=90,
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concurrency_limit=5,
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)
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stream.ui.launch()
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