pyannote
pyannote/speaker-diarization-3.1
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Model Documentation
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This pipeline is the same as
It ingests mono audio sampled at 16kHz and outputs speaker diarization as an
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.
1. Install
Pre-loading audio files in memory may result in faster processing:
Hooks are available to monitor the progress of the pipeline:
In case the number of speakers is known in advance, one can use the
One can also provide lower and/or upper bounds on the number of speakers using
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
| Benchmark | DER% | FA% | Miss% | Conf% | Expected output | File-level evaluation | | ------------------------------------------------------------------------------------------------------------------------------------------| ---------------------------------- | --------------------------- | ---------------------------------- | ----------------------------------- | -------------------------------------------------------------------------------------------------------------------------------------------------- | -------------------------------------------------------------------------------------------------------------------------------------------------- |
| AISHELL-4 | 12.2 | 3.8 | 4.4 | 4.0 | RTTM | eval |
| AliMeeting (_channel 1_) | 24.4 | 4.4 | 10.0 | 10.0 | RTTM | eval |
| AMI (_headset mix,_ _only_words_) | 18.8 | 3.6 | 9.5 | 5.7 | RTTM | eval |
| AMI (_array1, channel 1,_ _only_words)_ | 22.4 | 3.8 | 11.2 | 7.5 | RTTM | eval |
| AVA-AVD | 50.0 | 10.8 | 15.7 | 23.4 | RTTM | eval |
| DIHARD 3 (_Full_) | 21.7 | 6.2 | 8.1 | 7.3 | RTTM | eval |
| MSDWild | 25.3 | 5.8 | 8.0 | 11.5 | RTTM | eval |
| REPERE (_phase 2_) | 7.8 | 1.8 | 2.6 | 3.5 | RTTM | eval |
| VoxConverse (_v0.3_) | 11.3 | 4.1 | 3.4 | 3.8 | RTTM | eval |
🎹 Speaker diarization 3.1
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:Requirements
1. Install
pyannote.audio 3.1 with pip install pyannote.audio
2. Accept pyannote/segmentation-3.0 user conditions
3. Accept pyannote/speaker-diarization-3.1 user conditions
4. Create access token at hf.co/settings/tokens.Usage
python
instantiate the pipeline
from pyannote.audio import Pipeline
pipeline = Pipeline.from_pretrained(
"pyannote/speaker-diarization-3.1",
use_auth_token="HUGGINGFACE_ACCESS_TOKEN_GOES_HERE")
run the pipeline on an audio file
diarization = pipeline("audio.wav")
dump the diarization output to disk using RTTM format
with open("audio.rttm", "w") as rttm:
diarization.write_rttm(rttm)
Processing on GPU
pyannote.audio pipelines run on CPU by default.
You can send them to GPU with the following lines:python
import torch
pipeline.to(torch.device("cuda"))
Processing from memory
Pre-loading audio files in memory may result in faster processing:
python
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:
python
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:python
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:python
diarization = pipeline("audio.wav", min_speakers=2, max_speakers=5)
Benchmark
This pipeline has been benchmarked on a large collection of datasets.
Processing is fully automatic:
... with the least forgiving diarization error rate (DER) setup (named _"Full"_ in this paper):
| Benchmark | DER% | FA% | Miss% | Conf% | Expected output | File-level evaluation | | ------------------------------------------------------------------------------------------------------------------------------------------
Citations
bibtex
@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},
}
bibtex
@inproceedings{Bredin23,
author={Hervé Bredin},
title={{pyannote.audio 2.1 speaker diarization pipeline: principle, benchmark, and recipe}},
year=2023,
booktitle={Proc. INTERSPEECH 2023},
}
Files & Weights
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