Streaming Speech Synthesis Without the Trade-Offs: Meet Streamflow
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Why this matters: Current diffusion speech models need to see the entire audio sequence, making them too slow and memory-heavy for assistants, agents, or anything that needs instant voice responses. Causal masks sound robotic; chunking adds weird seams. Streaming TTS has been stuck with a quality–latency tradeoff.
The idea: StreamFlow restricts attention using sliding windows over blocks:
Each block can see W_b past blocks and W_f future blocks
Compute becomes roughly O(B × W × N) instead of full O(N²)
Prosody stays smooth, latency stays constant, and boundaries disappear with small overlaps + cross-fades
How it works: The system is still a Diffusion Transformer, but trained in two phases:
Full-attention pretraining for global quality
Block-wise fine-tuning to adapt to streaming constraints
Generates mel-spectrograms; BigVGAN vocoder runs in parallel.
Performance:
~180ms first-packet latency (80ms model, 60ms vocoder, 40ms overhead)
No latency growth with longer speech
MOS tests show near-indistinguishable quality vs non-streaming diffusion
Speaker similarity within ~2%, prosody continuity preserved
Key ablation takeaways:
Past context helps until ~3 blocks; more adds little
Even a tiny future window greatly boosts naturalness
Best results: 0.4–0.6s block size, ~10–20% overlap
Comparison:
Autoregressive TTS → streaming but meh quality
GAN TTS → fast but inconsistent
Causal diffusion → real-time but degraded
StreamFlow → streaming + near-SOTA quality
Bigger picture: Smart attention shaping lets diffusion models work in real time without throwing away global quality. The same technique could apply to streaming music generation, translation, or interactive media.