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Toolkit to benchmark various speech recognition APIs (NeMo, Whisper...) and visualize the results

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ASR Benchmark

Toolkit to benchmark various speech recognition APIs (NeMo, Whisper...) and visualize the results. Supported models are mostly french. It can compute WER, RTF (or latencies when streaming) and measure hardware usage.

How to bench

Just run:

python benchmarker.py CONFIG_FILE

Data

The input data file (manifest) is a jsonl file (one json per line). Each line must have these fields:

  • "audio_filepath", the path to the audio file
  • "text", the text associated with the segment

They can also have:

  • "offset", the start of the segment in the audio (if not specified, it is equal to 0)
  • "duration", the duration of the segment (if not specified, the whole audio is used)
  • "name" or "dataset", the name of the dataset

Examples

Examples are provided in the examples folder. There is a audio file to test with benchmark config file, and a notebook for generating plots.

Requirements

You need to install the package so that benchmarker.py can find the source code:

pip install -e .

You also need to install ssak repo. You can run:

pip install git+https://github.com/linagora-labs/ssak

Then depending on what you want to bench, you will need to install other packages like faster-whisper, nemo(nemo_toolkit['asr']), whisper and transformers.

Tools

Some tools are avaialble in the tools folder:

  • add_silence.py: a script for adding white noise to audio files
  • subsample_data.py: for selecting a subset of specified datasets

Don't hesitate to submit your tools (for converting datasets to the jsonl format for example). I used scripts from ssak to do it but datasets were in kaldi format.

Backends (interfaces)

The current available backends:

  • HTTP-API ("http-api")
  • LinTo-STT ("linto-stt"): for using whisper, kaldi or nemo models. Can be streaming (can compute latencies) or offline
  • Whisper ("openai")
  • Faster Whisper ("faster-whisper")
  • Transformers ("transformers"): work for Whisper
  • Transformers Intel ("intel-transformers"): for using intel extension
  • Transformers Facebook ("transformers-facebook"): for MMS model
  • Transformers Bofenghuang ("transformers-bofenghuang"): for the french finetuned wav2vec
  • NeMo ("nemo")

If the available interfaces don't allow to bench a model you want, you can easily add it by folliwing these steps:

  • You create new class that inherits from asr_benchmark.benchmark.interfaces.Model
  • You implement the various functions (load, transcribe, ...)
  • You add your backend in asr_benchmark.benchmark.backend_to_model

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Toolkit to benchmark various speech recognition APIs (NeMo, Whisper...) and visualize the results

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