🚀 Speaker diarization 3.1
This open - source pipeline focuses on speaker diarization. It solves the problem of accurately identifying different speakers in an audio file. Compared with its predecessor, it removes the problematic use of onnxruntime
, running both speaker segmentation and embedding in pure PyTorch, which eases deployment and may speed up inference.
🚀 Quick Start
Using this open - source model in production? Consider switching to pyannoteAI for better and faster options.
This pipeline is the same as pyannote/speaker-diarization-3.0
except it removes the problematic use of onnxruntime
. Both speaker segmentation and embedding now run in pure PyTorch. This should ease deployment and possibly speed up inference. It requires pyannote.audio version 3.1 or higher.
It ingests mono audio sampled at 16kHz and outputs speaker diarization as an Annotation
instance:
- Stereo or multi - channel audio files are automatically downmixed to mono by averaging the channels.
- Audio files sampled at a different rate are resampled to 16kHz automatically upon loading.
✨ Features
- Pure PyTorch: Runs speaker segmentation and embedding in pure PyTorch, removing the use of
onnxruntime
.
- Automatic Pre - processing: Automatically downmixes multi - channel audio to mono and resamples audio to 16kHz.
- Benchmarked Performance: Has been benchmarked on a large collection of datasets with a strict DER setup.
📦 Installation
- Install
pyannote.audio
3.1
with pip install pyannote.audio
- Accept
pyannote/segmentation-3.0
user conditions
- Accept
pyannote/speaker-diarization-3.1
user conditions
- Create access token at
hf.co/settings/tokens
.
💻 Usage Examples
Basic Usage
from pyannote.audio import Pipeline
pipeline = Pipeline.from_pretrained(
"pyannote/speaker-diarization-3.1",
use_auth_token="HUGGINGFACE_ACCESS_TOKEN_GOES_HERE")
diarization = pipeline("audio.wav")
with open("audio.rttm", "w") as rttm:
diarization.write_rttm(rttm)
Advanced Usage
Processing on GPU
pyannote.audio
pipelines run on CPU by default. You can send them to GPU with the following lines:
import torch
pipeline.to(torch.device("cuda"))
Processing from memory
Pre - loading audio files in memory may result in faster processing:
waveform, sample_rate = torchaudio.load("audio.wav")
diarization = pipeline({"waveform": waveform, "sample_rate": sample_rate})
Monitoring progress
Hooks are available to monitor the progress of the pipeline:
from pyannote.audio.pipelines.utils.hook import ProgressHook
with ProgressHook() as hook:
diarization = pipeline("audio.wav", hook=hook)
Controlling the number of speakers
In case the number of speakers is known in advance, one can use the num_speakers
option:
diarization = pipeline("audio.wav", num_speakers=2)
One can also provide lower and/or upper bounds on the number of speakers using min_speakers
and max_speakers
options:
diarization = pipeline("audio.wav", min_speakers=2, max_speakers=5)
📚 Documentation
This pipeline has been benchmarked on a large collection of datasets. Processing is fully automatic:
- No manual voice activity detection (as is sometimes the case in the literature)
- No manual number of speakers (though it is possible to provide it to the pipeline)
- No fine - tuning of the internal models nor tuning of the pipeline hyper - parameters to each dataset
... with the least forgiving diarization error rate (DER) setup (named "Full" in this paper):
- No forgiveness collar
- Evaluation of overlapped speech
📄 License
This project is licensed under the MIT license.
📚 Citations
@inproceedings{Plaquet23,
author={Alexis Plaquet and Hervé Bredin},
title={{Powerset multi-class cross entropy loss for neural speaker diarization}},
year=2023,
booktitle={Proc. INTERSPEECH 2023},
}
@inproceedings{Bredin23,
author={Hervé Bredin},
title={{pyannote.audio 2.1 speaker diarization pipeline: principle, benchmark, and recipe}},
year=2023,
booktitle={Proc. INTERSPEECH 2023},
}