Streaming speaker diarization
Overlap-aware low-latency online speaker diarization based on end-to-end local segmentation
by Juan Manuel Coria, Hervé Bredin, Sahar Ghannay and Sophie Rosset.
We propose to address online speaker diarization as a combination of incremental clustering and local diarization applied to a rolling buffer updated every 500ms. Every single step of the proposed pipeline is designed to take full advantage of the strong ability of a recently proposed end-to-end overlap-aware segmentation to detect and separate overlapping speakers. In particular, we propose a modified version of the statistics pooling layer (initially introduced in the x-vector architecture) to give less weight to frames where the segmentation model predicts simultaneous speakers. Furthermore, we derive cannot-link constraints from the initial segmentation step to prevent two local speakers from being wrongfully merged during the incremental clustering step. Finally, we show how the latency of the proposed approach can be adjusted between 500ms and 5s to match the requirements of a particular use case, and we provide a systematic analysis of the influence of latency on the overall performance (on AMI, DIHARD and VoxConverse).
Citation
Paper currently under review.
Installation
- Create environment:
conda create -n diarization python==3.8
conda activate diarization
-
Install the latest PyTorch version following the official instructions
-
Install dependencies:
pip install -r requirements.txt
Usage
CLI
Stream a previously recorded conversation:
python main.py /path/to/audio.wav
Or use a real audio stream from your microphone:
python main.py microphone
This will launch a real-time visualization of the diarization outputs as they are produced by the system:
By default, the script uses step = latency = 500ms, and it sets reasonable values for all hyper-parameters. See python main.py -h
for more information.
API
We provide various building blocks that can be combined to process an audio stream. Our streaming implementation is based on RxPY, but the functional
module is completely independent.
In this example we show how to obtain speaker embeddings from a microphone stream with Equation 2:
from sources import MicrophoneAudioSource
from functional import FrameWiseModel, ChunkWiseModel, OverlappedSpeechPenalty, EmbeddingNormalization
mic = MicrophoneAudioSource(sample_rate=16000)
# Initialize independent modules
segmentation = FrameWiseModel("pyannote/segmentation")
embedding = ChunkWiseModel("pyannote/embedding")
osp = OverlappedSpeechPenalty(gamma=3, beta=10)
normalization = EmbeddingNormalization(norm=1)
# Branch the microphone stream to calculate segmentation
segmentation_stream = mic.stream.pipe(ops.map(segmentation))
# Join audio and segmentation stream to calculate speaker embeddings
embedding_stream = rx.zip(mic.stream, segmentation_stream).pipe(
ops.starmap(lambda wave, seg: (wave, osp(seg))),
ops.starmap(embedding),
ops.map(normalization)
)
embedding_stream.suscribe(on_next=lambda emb: print(emb.shape))
mic.read()
Output:
(4, 512)
(4, 512)
(4, 512)
...
Reproducible research
In order to reproduce the results of the paper, use the following hyper-parameters:
Dataset | latency | tau | rho | delta |
---|---|---|---|---|
DIHARD III | any | 0.555 | 0.422 | 1.517 |
AMI | any | 0.507 | 0.006 | 1.057 |
VoxConverse | any | 0.576 | 0.915 | 0.648 |
DIHARD II | 1s | 0.619 | 0.326 | 0.997 |
DIHARD II | 5s | 0.555 | 0.422 | 1.517 |
For instance, for a DIHARD III configuration, one would use:
python main.py /path/to/file.wav --latency=5 --tau=0.555 --rho=0.422 --delta=1.517 --output /output/dir
And then to obtain the diarization error rate:
from pyannote.metrics.diarization import DiarizationErrorRate
from pyannote.database.util import load_rttm
metric = DiarizationErrorRate()
hypothesis = load_rttm("/output/dir/output.rttm")
hypothesis = list(hypothesis.values())[0] # Extract hypothesis from dictionary
reference = load_rttm("/path/to/reference.rttm")
reference = list(reference.values())[0] # Extract reference from dictionary
der = metric(reference, hypothesis)
For convenience and to facilitate future comparisons, we also provide the expected outputs in RTTM format corresponding to every entry of Table 1 and Figure 5 in the paper. This includes the VBx offline baseline as well as our proposed online approach with latencies 500ms, 1s, 2s, 3s, 4s, and 5s.
License
MIT License
Copyright (c) 2021 Université Paris-Saclay
Copyright (c) 2021 CNRS
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