GEPARD - Generative, Prosody-aware, Autoregressive text-to-speech model for Realtime Dialogue
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Abstract (arXiv)
We present GEPARD (Generative, Prosody-aware, Autoregressive text-to-speech model for Realtime Dialogue), a streaming text-to-speech model for real-time spoken dialogue. GEPARD generates speech autoregressively with an LLM backbone - text and audio embeddings are trained together in a single decoder-only model - and decodes it to a waveform with an FSQ-based neural codec, streaming audio chunk-by-chunk as text arrives. Our central goal is a TTS architecture served by a standard LLM engine (vLLM) without modifying its compute kernels. This defines the overarching design principle: the backbone is a standard full-attention transformer, while all non-trivial auxiliary mechanisms - zero-shot voice cloning, text augmentation, and classifier-free guidance - are moved out of the autoregressive decode loop into prefill, or distilled directly into the weights. On streaming end-to-end inference, a single stream reaches a Real-Time Factor of about 0.067 (roughly 15x faster than real-time); under 256 concurrent streams the system reaches an aggregate speedup of about 204x on a single server-class GPU. We detail: (1) system-level solutions for vLLM-native serving; (2) the "short register" (1-2 word) failure mode of autoregressive speech decoders, with diagnostic probes and a mitigation; and (3) distillation of two-pass classifier-free guidance over text into single-pass weights via Direct Preference Optimization (DPO).
Key facts. No citing papers in the map yet (data through 14 Oct 2026). Builds on 13 of its 23 references that are in the map. First posted on arXiv 4 Jul 2026.
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References · 13
- 2023Efficient Memory Management for Large Language Model Serving with PagedAttentioncited by 5,944
- 2022Flamingo: a Visual Language Model for Few-Shot Learningcited by 4,386
- 2020Supervised Contrastive Learningcited by 2,829
- 2023FlashAttention-2: Faster Attention with Better Parallelism and Work Partitioningcited by 2,598
- 2021WavLM: Large-Scale Self-Supervised Pre-Training for Full Stack Speech Processingcited by 1,657
- 2021VICReg: Variance-Invariance-Covariance Regularization for Self-Supervised Learningcited by 1,057
- 2023Neural Codec Language Models are Zero-Shot Text to Speech Synthesizerscited by 752
- 2021Perceiver IO: A General Architecture for Structured Inputs & Outputscited by 629
- 2024Moshi: a speech-text foundation model for real-time dialoguecited by 568
- 2022UTMOS: UTokyo-SaruLab System for VoiceMOS Challenge 2022cited by 390
- 2022Tensor Programs V: Tuning Large Neural Networks via Zero-Shot Hyperparameter Transfercited by 288
- 2024VALL-E 2: Neural Codec Language Models are Human Parity Zero-Shot Text to Speech Synthesizerscited by 122
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- 2025From Text to Talk: Audio-Language Model Needs Non-Autoregressive Joint Training
- 2026CuteTTS: Efficient and High-Quality Speech Synthesis via Autoregressive Modeling of Continuous Latents
- 2026FlashTTS: Fast Streaming TTS with MTP Acceleration and X-pred Mean Flow Distillation
- 2025LLMVoX: Autoregressive Streaming Text-to-Speech Model for Any LLM
- 2025ZipVoice-Dialog: Non-Autoregressive Spoken Dialogue Generation with Flow Matching
- 2026Controlling Speaking Rate in Autoregressive TTS via Activation Steering
- 2023Speak While You Think: Streaming Speech Synthesis During Text Generation
- 2026TLDR: Compressing Audio Tokens for Efficient Autoregressive Text-to-Speech