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whisper_streaming

Real-time streaming transcription and translation for Whisper models.

Real-Time Whisper Streaming

Turn OpenAI Whisper into a low-latency speech-to-text and translation system.

Free forever · open source · MIT
✓ Light setup✓ Browser-based✓ Real-time collaboration
6.5
ZEKAI
Low-latency streaming Whisper
Free & open source · Government
Ease of setup5
Usefulness8
Maintenance5.5
Community7.5
★ 3,679
GITHUB
Nov 2025
UPDATED
MIT
LICENSE
🛠 Don't want to self-host? We can set it up →
⚡ Quick answer

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.

Runs on
Cross-platform
Hosting
Self-host
Languages
Python
License
MIT

What it is

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.

Who it's for

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.

Key features

Multiple Backends

Supports faster-whisper, whisper-timestamped, OpenAI API, and Apple Silicon MLX backends.

Voice Activity Detection

Integrates optional VAD and voice activity controllers using PyTorch.

Buffer Trimming

Offers segment or sentence-level buffer trimming options to manage latency and quality.

Real-Time Simulation

Includes scripts to simulate live streaming directly from local 16kHz mono WAV files.

How to start

Up and running in 3 steps

1
Install audio processing libraries using pip install librosa soundfile.
2
Install your chosen Whisper backend, such as faster-whisper with NVIDIA CUDA support.
3
Run the transcription simulation script via whisper_online.py with your audio path.
Real talk

Our honest verdict

What's great

  • Achieves low latency around 3.3 seconds on unsegmented long-form speech.
  • Flexible backend choice including GPU-accelerated and local Apple Silicon options.
  • Optional voice activity controllers and detectors reduce processing overhead.
  • Built-in support for live translation tasks alongside transcription.

Where it falls short

  • Note that the project has been succeeded by SimulStreaming in 2025.
  • Requires technical setup of Python environments, audio libraries, and optional GPU drivers.
  • Using the OpenAI API backend incurs ongoing per-request API costs that scale with chunk size.
Cost

Pricing & license

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.

Paid work

Open remote AI-training roles

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.

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Questions

FAQ

What backends are supported?

It supports faster-whisper, whisper-timestamped, OpenAI API, and Whisper MLX for Apple Silicon.

Is a GPU required?

A GPU with CUDA support is recommended for faster-whisper, but you can also use Apple Silicon MLX or the OpenAI API backend.

How does it handle latency?

It uses a local agreement policy with self-adaptive latency and minimum chunk sizing to process audio in real time.

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