KIFF/pyannote-speaker-diarization-endpoint

🎹 Speaker diarization

Relies on pyannote.audio 2.0: see installation instructions.

TL;DR

# load the pipeline from Hugginface Hub
from pyannote.audio import Pipeline
pipeline = Pipeline.from_pretrained("pyannote/[email protected]")

# apply the pipeline to 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)

Advanced usage

If the number of speakers is known in advance, you can include the num_speakers parameter in the parameters dictionary:

handler = EndpointHandler()
diarization = handler({"inputs": base64_audio, "parameters": {"num_speakers": 2}})

You can also provide lower and/or upper bounds on the number of speakers using the min_speakers and max_speakers parameters:

handler = EndpointHandler()
diarization = handler({"inputs": base64_audio, "parameters": {"min_speakers": 2, "max_speakers": 5}})

If you're feeling adventurous, you can experiment with various pipeline hyperparameters. For instance, you can use a more aggressive voice activity detection by increasing the value of segmentation_onset threshold:

hparams = handler.pipeline.parameters(instantiated=True)
hparams["segmentation_onset"] += 0.1
handler.pipeline.instantiate(hparams)

To apply the updated handler for the API inference that can handle the number of speakers, use the following code:

from typing import Dict
from pyannote.audio import Pipeline
import torch 
import base64
import numpy as np

SAMPLE_RATE = 16000

class EndpointHandler():
    def __init__(self, path=""):
        # load the model
        self.pipeline = Pipeline.from_pretrained("KIFF/pyannote-speaker-diarization-endpoint")

    def __call__(self, data: Dict[str, bytes]) -> Dict[str, str]:
        """
        Args:
            data (:obj:):
                includes the deserialized audio file as bytes
        Return:
            A :obj:`dict`:. base64 encoded image
        """
        # process input
        inputs = data.pop("inputs", data)
        parameters = data.pop("parameters", None) #  min_speakers=2, max_speakers=5

        # decode the base64 audio data
        audio_data = base64.b64decode(inputs)
        audio_nparray = np.frombuffer(audio_data, dtype=np.int16)

        # prepare pynannote input
        audio_tensor= torch.from_numpy(audio_nparray).float().unsqueeze(0)
        pyannote_input = {"waveform": audio_tensor, "sample_rate": SAMPLE_RATE}
        
        # apply pretrained pipeline
        # pass inputs with all kwargs in data
        if parameters is not None:
            diarization = self.pipeline(pyannote_input, **parameters)
        else:
            diarization = self.pipeline(pyannote_input)

        # postprocess the prediction
        processed_diarization = [
            {"label": str(label), "start": str(segment.start), "stop": str(segment.end)}
            for segment, _, label in diarization.itertracks(yield_label=True)
        ]
        
        return {"diarization": processed_diarization}

Benchmark

Real-time factor

Real-time factor is around 5% using one Nvidia Tesla V100 SXM2 GPU (for the neural inference part) and one Intel Cascade Lake 6248 CPU (for the clustering part).

In other words, it takes approximately 3 minutes to process a one hour conversation.

Accuracy

This pipeline is benchmarked on a growing 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	14.61	3.31	4.35	6.95	RTTM	eval
AMI Mix-Headset only_words	18.21	3.28	11.07	3.87	RTTM	eval
AMI Array1-01 only_words	29.00	2.71	21.61	4.68	RTTM	eval
CALLHOME Part2	30.24	3.71	16.86	9.66	RTTM	eval
DIHARD 3 Full	20.99	4.25	10.74	6.00	RTTM	eval
REPERE Phase 2	12.62	1.55	3.30	7.76	RTTM	eval
VoxConverse v0.0.2	12.76	3.45	3.85	5.46	RTTM	eval

Support

For commercial enquiries and scientific consulting, please contact me.
For technical questions and bug reports, please check pyannote.audio Github repository.

Citations

@inproceedings{Bredin2021,
  Title = {{End-to-end speaker segmentation for overlap-aware resegmentation}},
  Author = {{Bredin}, Herv{\'e} and {Laurent}, Antoine},
  Booktitle = {Proc. Interspeech 2021},
  Address = {Brno, Czech Republic},
  Month = {August},
  Year = {2021},
}

@inproceedings{Bredin2020,
  Title = {{pyannote.audio: neural building blocks for speaker diarization}},
  Author = {{Bredin}, Herv{\'e} and {Yin}, Ruiqing and {Coria}, Juan Manuel and {Gelly}, Gregory and {Korshunov}, Pavel and {Lavechin}, Marvin and {Fustes}, Diego and {Titeux}, Hadrien and {Bouaziz}, Wassim and {Gill}, Marie-Philippe},
  Booktitle = {ICASSP 2020, IEEE International Conference on Acoustics, Speech, and Signal Processing},
  Address = {Barcelona, Spain},
  Month = {May},
  Year = {2020},
}