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

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},
}