Update README.md

README.md

@@ -5,4 +5,130 @@ pipeline_tag: text-to-speech
 
 This repository contains the model as described in [LLMVoX: Autoregressive Streaming Text-to-Speech Model for Any LLM](https://hf.co/papers/2503.04724).
 
-For more information, check out the project page at https://mbzuai-oryx.github.io/LLMVoX/ and the code at https://github.com/mbzuai-oryx/LLMVoX.
+For more information, check out the project page at https://mbzuai-oryx.github.io/LLMVoX/ and the code at https://github.com/mbzuai-oryx/LLMVoX.
+
+# LLMVoX: Autoregressive Streaming Text-to-Speech Model for Any LLM
+
+<div>
+<a href="https://mbzuai-oryx.github.io/LLMVoX/"><img src="https://img.shields.io/badge/Project-Page-blue" alt="Project Page"></a>
+<a href="https://arxiv.org/abs/2503.04724"><img src="https://img.shields.io/badge/arXiv-2503.04724-b31b1b.svg" alt="arXiv"></a>
+<a href="https://github.com/mbzuai-oryx/LLMVoX/"><img src="https://img.shields.io/badge/GitHub-LLMVoX-black?logo=github" alt="GitHub Repository"></a>
+<a href="https://github.com/mbzuai-oryx/LLMVoX/blob/main/LICENSE"><img src="https://img.shields.io/badge/License-MIT-yellow.svg" alt="License: MIT"></a>
+</div>
+
+**Authors:**  
+**[Sambal Shikar](https://github.com/mbzuai-oryx/LLMVoX?tab=readme-ov-file)**, **[Mohammed Irfan K](https://scholar.google.com/citations?user=GJp0keYAAAAJ&hl=en)**, **[Sahal Shaji Mullappilly](https://scholar.google.com/citations?user=LJWxVpUAAAAJ&hl=en)**, **[Fahad Khan](https://sites.google.com/view/fahadkhans/home)**, **[Jean Lahoud](https://scholar.google.com/citations?user=LsivLPoAAAAJ&hl=en)**, **[Rao Muhammad Anwer](https://scholar.google.com/citations?hl=en&authuser=1&user=_KlvMVoAAAAJ)**, **[Salman Khan](https://salman-h-khan.github.io/)**, **[Hisham Cholakkal](https://scholar.google.com/citations?hl=en&user=bZ3YBRcAAAAJ)**
+
+**Mohamed Bin Zayed University of Artificial Intelligence (MBZUAI), UAE**
+
+<p align="center">
+    <img src="assets/arch_diagram.svg" alt="LLMVoX Architecture" width="800px">
+</p>
+
+## Overview
+
+LLMVoX is a lightweight 30M-parameter, LLM-agnostic, autoregressive streaming Text-to-Speech (TTS) system designed to convert text outputs from Large Language Models into high-fidelity streaming speech with low latency.
+
+Key features:
+- 🚀 **Lightweight & Fast**: Only 30M parameters with end-to-end latency as low as 300ms
+- 🔌 **LLM-Agnostic**: Works with any LLM and Vision-Language Model without fine-tuning
+- 🌊 **Multi-Queue Streaming**: Enables continuous, low-latency speech generation
+- 🌐 **Multilingual Support**: Adaptable to new languages with dataset adaptation
+
+## Quick Start
+
+### Installation
+
+```bash
+# Requirements: CUDA 11.7+, Flash Attention 2.0+ compatible GPU
+
+git clone https://github.com/mbzuai-oryx/LLMVoX.git
+cd LLMVoX
+
+conda create -n llmvox python=3.9
+conda activate llmvox
+
+pip3 install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu118
+pip install flash-attn --no-build-isolation
+pip install -r requirements.txt
+
+# Download checkpoints from Hugging Face
+# https://huggingface.co/MBZUAI/LLMVoX/tree/main
+mkdir -p CHECKPOINTS
+# Download wavtokenizer_large_speech_320_24k.ckpt and ckpt_english_tiny.pt
+```
+
+### Voice Chat
+
+```bash
+# Basic usage
+python streaming_server.py --chat_type voice --llm_checkpoint "meta-llama/Llama-3.1-8B-Instruct"
+
+# With multiple GPUs
+python streaming_server.py --chat_type voice --llm_checkpoint "meta-llama/Llama-3.1-8B-Instruct" \
+  --llm_device "cuda:0" --tts_device_1 1 --tts_device_2 2
+
+# Balance latency/quality
+python streaming_server.py --chat_type voice --llm_checkpoint "meta-llama/Llama-3.1-8B-Instruct" \
+  --initial_dump_size_1 10 --initial_dump_size_2 160 --max_dump_size 1280
+```
+
+### Text Chat & Visual Speech
+
+```bash
+# Text-to-Speech
+python streaming_server.py --chat_type text --llm_checkpoint "meta-llama/Llama-3.1-8B-Instruct"
+
+# Visual Speech (Speech + Image → Speech)
+python streaming_server.py --chat_type visual_speech --llm_checkpoint "Qwen/Qwen2.5-VL-7B-Instruct" \
+  --eos_token "<|im_end|>"
+
+# Multimodal (support for models like Phi-4)
+python streaming_server.py --chat_type multimodal --llm_checkpoint "microsoft/Phi-4-multimodal-instruct" \
+  --eos_token "<|end|>"
+```
+
+## API Reference
+
+| Endpoint | Purpose | Required Parameters |
+|----------|---------|---------------------|
+| `/tts` | Text-to-speech | `text`: String to convert |
+| `/voicechat` | Voice conversations | `audio_base64`, `source_language`, `target_language` |
+| `/multimodalchat` | Voice + multiple images | `audio_base64`, `image_list` |
+| `/vlmschat` | Voice + single image | `audio_base64`, `image_base64`, `source_language`, `target_language` |
+
+## Local UI Demo
+
+<p align="center">
+    <img src="assets/ui.png" alt="Demo UI" width="800px">
+</p>
+
+```bash
+# Start server
+python streaming_server.py --chat_type voice --llm_checkpoint "meta-llama/Llama-3.1-8B-Instruct" --api_port PORT
+
+# Launch UI
+python run_ui.py --ip STREAMING_SERVER_IP --port PORT
+```
+
+## Citation
+
+```bibtex
+@article{shikhar2025llmvox,
+  title={LLMVoX: Autoregressive Streaming Text-to-Speech Model for Any LLM},
+  author={Shikhar, Sambal and Kurpath, Mohammed Irfan and Mullappilly, Sahal Shaji and Lahoud, Jean and Khan, Fahad and Anwer, Rao Muhammad and Khan, Salman and Cholakkal, Hisham},
+  journal={arXiv preprint arXiv:2503.04724},
+  year={2025}
+}
+```
+
+## Acknowledgments
+
+- [Andrej Karpathy's NanoGPT](https://github.com/karpathy/nanoGPT)
+- [WavTokenizer](https://github.com/jishengpeng/WavTokenizer)
+- [Whisper](https://github.com/openai/whisper)
+- [Neural G2P](https://github.com/lingjzhu/CharsiuG2P)
+
+## License
+
+This project is licensed under the MIT License - see the [LICENSE](LICENSE) file for details.