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AUTHOR Hervé Bredin - http://herve.niderb.fr
In this tutorial, you will learn how to train a speech turn embedding using pyannote-speaker-embedding
command line tool.
If you use pyannote-audio
for speaker (or audio) neural embedding, please cite the following paper:
@inproceedings{Bredin2017,
author = {Herv\'{e} Bredin},
title = {{TristouNet: Triplet Loss for Speaker Turn Embedding}},
booktitle = {42nd IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2017},
year = {2017},
url = {http://arxiv.org/abs/1609.04301},
}
$ source activate pyannote
$ pip install pyannote.db.odessa.ami
$ pip install pyannote.db.musan
$ pip install pyannote.db.voxceleb
This tutorial relies on the VoxCeleb, AMI and MUSAN databases. We first need to tell pyannote
where the audio files are located:
$ cat ~/.pyannote/database.yml
Databases:
VoxCeleb: /path/to/voxceleb1/*/wav/{uri}.wav
AMI: /path/to/ami/amicorpus/*/audio/{uri}.wav
MUSAN: /path/to/musan/{uri}.wav
Have a look at pyannote.database
documentation to learn how to use other datasets.
To ensure reproducibility, pyannote-speaker-embedding
relies on a configuration file defining the experimental setup:
$ cat tutorials/models/speaker_embedding/config.yml
feature_extraction:
name: LibrosaMFCC
params:
e: False
De: True
DDe: True
coefs: 19
D: True
DD: True
duration: 0.025
step: 0.010
sample_rate: 16000
data_augmentation:
name: AddNoise
params:
snr_min: 10
snr_max: 20
collection: MUSAN.Collection.BackgroundNoise
architecture:
name: ClopiNet
params:
instance_normalize: True
rnn: LSTM
recurrent: [256, 256, 256]
linear: [256]
bidirectional: True
pooling: sum
batch_normalize: True
normalize: True
approach:
name: TripletLoss
params:
metric: cosine
clamp: sigmoid
margin: 0.0
min_duration: 0.500
max_duration: 1.500
sampling: all
per_fold: 20
per_label: 3
per_epoch: 1
label_min_duration: 60
scheduler:
name: CyclicScheduler
params:
epochs_per_cycle: 14
The following command will train the network using VoxCeleb1 for 1000 epochs (one epoch = one day of audio)
$ export EXPERIMENT_DIR=tutorials/models/speaker_embedding
$ pyannote-speaker-embedding train --gpu --to=1000 ${EXPERIMENT_DIR} VoxCeleb.SpeakerVerification.VoxCeleb1
This will create a bunch of files in TRAIN_DIR
(defined below).
One can follow along the training process using tensorboard.
$ tensorboard --logdir=${EXPERIMENT_DIR}
To get a quick idea of how the network is doing during training, one can use the validate
mode.
It can (should!) be run in parallel to training and evaluates the model epoch after epoch.
One can use tensorboard to follow the validation process.
$ export TRAIN_DIR=${EXPERIMENT_DIR}/train/VoxCeleb.SpeakerVerification.VoxCeleb1.train
$ pyannote-speaker-embedding validate --subset=test ${TRAIN_DIR} VoxCeleb.SpeakerDiarization.VoxCeleb1
This model reaches approximately 7% EER on VoxCeleb1.
Now that we know how the model is doing, we can apply it on all files of the AMI database store embeddings in /path/to/precomputed/emb
:
$ pyannote-speaker-embedding apply ${TRAIN_DIR}/weights/2000.pt AMI.SpeakerDiarization.MixHeadset /path/to/precomputed/emb
We can then use these extracted embeddings like this:
# first test file of AMI protocol
>>> from pyannote.database import get_protocol
>>> protocol = get_protocol('AMI.SpeakerDiarization.MixHeadset')
>>> test_file = next(protocol.test())
# precomputed embeddings as pyannote.core.SlidingWindowFeature
>>> from pyannote.audio.features import Precomputed
>>> precomputed = Precomputed('/path/to/precomputed/emb')
>>> embeddings = precomputed(test_file)
# iterate over all embeddings
>>> for window, embedding in embeddings:
... print(window)
... print(embedding)
... break
# extract embedding from a specific segment
>>> from pyannote.core import Segment
>>> fX = embeddings.crop(Segment(10, 20))
>>> print(fX.shape)
For more options, see:
$ pyannote-speaker-embedding --help
That's all folks!