SpeakStream: Streaming Text-to-Speech with Interleaved Data
Richard He Bai, Zijin Gu, Tatiana Likhomanenko, Navdeep Jaitly
TL;DR
SpeakStream addresses high first-token latency in cascaded LLM+TTS systems by introducing a decoder-only streaming TTS that interleaves text and speech into a single sequence and is trained with next-step prediction on interleaved data. It eliminates explicit alignment, uses kv-cache for efficiency, and pairs with VocStream to deliver end-to-end streaming with first-frame latency around $30\,\mathrm{ms}$ and total latency under $50\,\mathrm{ms}$ on a Mac Mini. Empirical results show competitive WER and human evaluations indicate maintained coherence compared with non-streaming baselines, demonstrating practical viability for interactive agents. The work highlights a unified, low-latency approach to streaming TTS that could underpin responsive conversational systems and cascaded AI pipelines.
Abstract
The latency bottleneck of traditional text-to-speech (TTS) systems fundamentally hinders the potential of streaming large language models (LLMs) in conversational AI. These TTS systems, typically trained and inferenced on complete utterances, introduce unacceptable delays, even with optimized inference speeds, when coupled with streaming LLM outputs. This is particularly problematic for creating responsive conversational agents where low first-token latency is critical. In this paper, we present SpeakStream, a streaming TTS system that generates audio incrementally from streaming text using a decoder-only architecture. SpeakStream is trained using a next-step prediction loss on interleaved text-speech data. During inference, it generates speech incrementally while absorbing streaming input text, making it particularly suitable for cascaded conversational AI agents where an LLM streams text to a TTS system. Our experiments demonstrate that SpeakStream achieves state-of-the-art latency results in terms of first-token latency while maintaining the quality of non-streaming TTS systems.
