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AI Audio2023
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faster-whisper

Reimplements OpenAI's Whisper speech-to-text on the CTranslate2 inference engine, running up to 4x faster at the same accuracy while using less memory. Adds a batched pipeline, 8-bit quantization, VAD filtering, and word-level timestamps.

Introduction

Whisper's accuracy was rarely the bottleneck — its runtime was. faster-whisper keeps the model weights byte-for-byte identical to OpenAI's Whisper but swaps the PyTorch runtime for CTranslate2, a quantization-aware inference engine. That single change transcribes a 13-minute clip in about 1 minute on GPU instead of ~2.5, with identical output.

What Sets It Apart
  • Same model, faster engine: because the weights are unchanged, you get OpenAI-quality transcripts at up to 4x the speed — there is no fine-tuning step or accuracy tradeoff to reason about.
  • Memory that fits smaller cards: INT8 quantization runs large-v2 in ~2.9 GB VRAM versus ~4.7 GB for openai/whisper, so it fits on consumer GPUs and runs usefully on CPU (small model: ~1m42s vs ~6m58s for 13 minutes of audio).
  • Throughput features built in: a batched inference pipeline, VAD filtering to skip silence, and word-level timestamps mean you rarely need to bolt on extra wrappers.
  • Distil-Whisper compatible: drop in distilled checkpoints for another speed step when you can trade a little accuracy.
Who It's For

Great fit if you run Whisper at volume — subtitling pipelines, meeting transcription, batch jobs — and want lower latency and memory without changing models or output quality. Look elsewhere if you need training or fine-tuning (this is inference-only), want a turnkey GUI app, or depend on PyTorch-specific Whisper hooks, since CTranslate2 is a separate runtime with its own model format.

Information

  • Websitegithub.com
  • AuthorsSYSTRAN
  • Published date2023/02/11

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