Real-time streaming transcription and translation for Whisper models.
Turn OpenAI Whisper into a low-latency speech-to-text and translation system.
Whisper-streaming provides real-time speech transcription and translation by building local agreement policy and self-adaptive latency on top of Whisper models. It suits developers and technologists building live audio processing pipelines for multilingual events.
The tool processes unsegmented long-form speech using multiple backends like faster-whisper, whisper-timestamped, OpenAI API, or Whisper MLX. It implements voice activity detection, chunking policies, and optional buffer trimming strategies to keep latency around 3.3 seconds.
It suits developers integrating real-time transcription into applications or live conference feeds. It does not suit end-users looking for a ready-to-use desktop application without command-line configuration.
Supports faster-whisper, whisper-timestamped, OpenAI API, and Apple Silicon MLX backends.
Integrates optional VAD and voice activity controllers using PyTorch.
Offers segment or sentence-level buffer trimming options to manage latency and quality.
Includes scripts to simulate live streaming directly from local 16kHz mono WAV files.
Free under the MIT license. If you choose to use the OpenAI API backend, you will need to pay OpenAI for API access costs based on your usage.
874 open remote AI-training roles, updated November 15, 2026; most pay $20–$170 an hour. Zekai lists them free; the application goes to the employer’s platform.
It supports faster-whisper, whisper-timestamped, OpenAI API, and Whisper MLX for Apple Silicon.
A GPU with CUDA support is recommended for faster-whisper, but you can also use Apple Silicon MLX or the OpenAI API backend.
It uses a local agreement policy with self-adaptive latency and minimum chunk sizing to process audio in real time.
See Zekai first in Google