kyutai-labs/delayed-streams-modeling

Kyutai's Speech-To-Text and Text-To-Speech models based on the Delayed Streams Modeling framework.

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kyutai-labs/delayed-streams-modeling has 3k+ stars and 300+ forks on GitHub, is written in Python, and was last pushed on Jan 26, 2026. Data as of . Signed-in members get hourly updates — create a free account.

Summary Information

Updated 11 minutes ago
Added to GitGenius on September 22nd, 2026
Created on June 16th, 2025
Open Issues & Pull Requests: 37 (+0)
GitHub issues: Enabled
Number of forks: 317
Total Stargazers: 3,041 (+0)
Total Subscribers: 31 (+0)

Repository Insights (GitGenius)

Median issue/PR response: 1.1 hours
Mean response time: 2.5 days
90th percentile: 5.3 days
Tracked items: 108

How this project is maintained

97% of open issues come from outside the core team, so the backlog reflects real-world use rather than internal planning. Three people close 70% of everything that gets resolved.

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Issue Activity (beta)

Open issues: 37
New in 7 days: 0
Closed in 7 days: 0
Avg open age: 351 days
Stale 30+ days: 36
Stale 90+ days: 31

Recent activity

Opened in 7 days: 0
Closed in 7 days: 0
Comments in 7 days: 0
Events in 7 days: 0

Top labels

  • question (42)
  • bug (13)
  • wontfix (1)

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Detailed Description

Delayed Streams Modeling is a framework for building speech-to-text and text-to-speech models that handle streaming audio and text tasks in real time.

The framework addresses the challenge of processing continuous audio or text streams with minimal latency while maintaining accuracy. It formalizes an approach used in prior models like Moshi and Hibiki, enabling efficient streaming inference across multiple modalities. The Kyutai STT models built on this framework process audio in chunks, return word-level timestamps, and include semantic voice activity detection in the smaller variant, making them suitable for interactive voice applications.

The speech-to-text models come in two sizes: a 1 billion parameter bilingual model supporting English and French with 0.5 second latency, and a 2.6 billion parameter English-only model with 2.5 second latency. Both support real-time streaming and efficient batching. Developers should choose this tool if they need production-grade speech recognition with streaming capabilities and word-level timing information. The project provides three separate implementations tailored to different deployment contexts: PyTorch for research and experimentation, Rust for production servers with websocket streaming, and MLX for on-device inference on Apple hardware. The Rust server implementation can serve 64 simultaneous connections on consumer GPU hardware at real-time performance factors.

Development activity shows consistent focus on practical deployment scenarios. The project maintains multiple implementation paths rather than a single codebase, indicating responsiveness to different user needs across research, production, and edge deployment. Documentation includes interactive notebooks and clear guidance on selecting the appropriate implementation for specific use cases. The team has built a complete voice AI system called Unmute using these models, demonstrating confidence in the framework's production readiness.