koljab/realtimestt

A robust, efficient, low-latency speech-to-text library with advanced voice activity detection, wake word activation and instant transcription.

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Summary Information

Updated 16 minutes ago
Added to GitGenius on September 6th, 2026
Created on August 29th, 2023
Open Issues & Pull Requests: 148 (+0)
GitHub issues: Enabled
Number of forks: 856
Total Stargazers: 10,115 (+0)
Total Subscribers: 74 (+0)

Repository Insights (GitGenius)

Median issue/PR response: 2.8 hours
Mean response time: 13.2 days
90th percentile: 18.6 days
Tracked items: 135

How this project is maintained

Around half of the issues opened in the past year never receive a reply. 100% of open issues come from outside the core team, so the backlog reflects real-world use rather than internal planning. Only 9% of issues opened in the past year have been closed.

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Issue Activity (beta)

Open issues: 106
New in 7 days: 0
Closed in 7 days: 0
Avg open age: 524 days
Stale 30+ days: 104
Stale 90+ days: 93

Recent activity

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

Top labels

  • duplicate (4)

Most active issues this week

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Detailed Description

RealtimeSTT is a Python speech-to-text library that provides voice activity detection, fast transcription, wake word activation, and direct audio stream access for real-time applications.

The library addresses the need for low-latency speech recognition in assistants, dictation tools, and streaming servers. It uses faster_whisper as its general-purpose default engine but supports multiple transcription backends through optional dependencies. The architecture distinguishes between streaming transcription for real-time text updates and final transcript refinement. For CPU-based production streaming on Linux x86-64, the tool recommends pairing sherpa-onnx-nemotron for fast incremental updates with sherpa-onnx-parakeet for authoritative final transcripts, processing only new audio frames during active speech turns rather than repeatedly retranscribing growing buffers. The library also includes native support for kroko-onnx as a local streaming ASR engine and silero-onnx-cpu for voice activity detection.

Developers should choose this tool if they need to integrate speech recognition into Python applications with minimal setup. It suits projects requiring microphone input handling, external audio stream processing, or browser-based streaming servers. The library is designed to work with only a few lines of code for basic use cases, while supporting advanced configurations for production deployments. The recommended engine profiles differ by hardware: CUDA setups continue using faster_whisper, while CPU deployments benefit from the two-engine approach that balances latency and quality.

The project maintains active development with regular updates to engine support and documentation. The maintainers have prioritized production deployment guidance through dedicated server setup documentation and pinned model configurations. Integration of community-contributed engines like kroko-onnx demonstrates responsiveness to user needs for specialized ASR solutions. The codebase targets Python 3.11 and 3.12 with documented platform-specific installation requirements, and the project uses multiprocessing for model operations with explicit guidance on script structure for cross-platform compatibility.