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app.py
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@@ -5,6 +5,7 @@ Based on the original liquid-audio demo
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from queue import Queue
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from threading import Thread
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import gradio as gr
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import numpy as np
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@@ -13,6 +14,25 @@ from fastrtc import AdditionalOutputs, ReplyOnPause, WebRTC
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from liquid_audio import ChatState, LFM2AudioModel, LFM2AudioProcessor, LFMModality
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# Load models
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HF_REPO = "LiquidAI/LFM2-Audio-1.5B"
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@@ -163,6 +183,7 @@ with gr.Blocks(title="LFM2-Audio Real-time Speech-to-Speech") as demo:
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modality="audio",
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mode="send-receive",
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full_screen=False,
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)
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with gr.Row():
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from queue import Queue
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from threading import Thread
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import os
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import gradio as gr
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import numpy as np
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from liquid_audio import ChatState, LFM2AudioModel, LFM2AudioProcessor, LFMModality
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# Configure WebRTC with STUN/TURN servers
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# This is CRITICAL for WebRTC connections to work through firewalls/NAT
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rtc_configuration = {
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"iceServers": [
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{
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"urls": [
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"stun:stun.l.google.com:19302",
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"stun:stun1.l.google.com:19302",
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]
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}
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]
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}
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# For production deployment on Hugging Face Spaces, you can use Cloudflare TURN:
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# Uncomment these lines and set TURN_KEY_ID and TURN_KEY_API_TOKEN as Secrets
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# from fastrtc import get_cloudflare_turn_credentials_async
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# if os.getenv("TURN_KEY_ID") and os.getenv("TURN_KEY_API_TOKEN"):
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# rtc_configuration = get_cloudflare_turn_credentials_async()
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# Load models
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HF_REPO = "LiquidAI/LFM2-Audio-1.5B"
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modality="audio",
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mode="send-receive",
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full_screen=False,
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rtc_configuration=rtc_configuration,
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)
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with gr.Row():
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