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Interleaved Speech-Text Language Models for Simple Streaming Text-to-Speech Synthesis
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This paper introduces Interleaved Speech-Text Language Model (IST-LM) for zero-shot streaming Text-to-Speech (TTS). Unlike many previous approaches, IST-LM is directly trained on interleaved sequences of text and speech tokens with a fixed ratio, eliminating the need for additional efforts like forced alignment or complex designs. The ratio of text chunk size to speech chunk size is crucial for the performance of IST-LM. To explore this, we conducted a comprehensive series of statistical analyses on the training data and performed correlation analysis with the final performance, uncovering several key factors: 1) the distance between speech tokens and their corresponding text tokens, 2) the number of future text tokens accessible to each speech token, and 3) the frequency of speech tokens precedes their corresponding text tokens. Experimental results demonstrate how to achieve an optimal streaming TTS system with a limited performance gap compared to its non-streaming counterpart. IST-LM is conceptually simple and empirically powerful, enabling streaming TTS with minimal overhead while largely preserving performance, and offering broad potential for integration with real-time text streams from large language models.
Forward citations
Cited by 3 Pith papers
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StreamMel: Real-Time Zero-shot Text-to-Speech via Interleaved Continuous Autoregressive Modeling
StreamMel interleaves text tokens with continuous mel frames in one autoregressive model, reaching state-of-the-art streaming latency with quality comparable to offline zero-shot TTS on LibriSpeech.
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SpeakStream: Streaming Text-to-Speech with Interleaved Data
A decoder-only TTS trained on force-aligned interleaved text-speech chunks generates audio after a few words, achieving ~30ms TTS latency and WER comparable to non-streaming.
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LLM as Effective Streaming Processor: Bridging Streaming-Batch Mismatches with Group Position Encoding
Streaming LLMs need only fix the input-attention mismatch; group position encoding (source and target positions numbered separately) removes the need for re-encoding and outperforms specialized streaming baselines.
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