2,621 research outputs found
Improving Source Separation via Multi-Speaker Representations
Lately there have been novel developments in deep learning towards solving
the cocktail party problem. Initial results are very promising and allow for
more research in the domain. One technique that has not yet been explored in
the neural network approach to this task is speaker adaptation. Intuitively,
information on the speakers that we are trying to separate seems fundamentally
important for the speaker separation task. However, retrieving this speaker
information is challenging since the speaker identities are not known a priori
and multiple speakers are simultaneously active. There is thus some sort of
chicken and egg problem. To tackle this, source signals and i-vectors are
estimated alternately. We show that blind multi-speaker adaptation improves the
results of the network and that (in our case) the network is not capable of
adequately retrieving this useful speaker information itself
Structured Sparsity Models for Multiparty Speech Recovery from Reverberant Recordings
We tackle the multi-party speech recovery problem through modeling the
acoustic of the reverberant chambers. Our approach exploits structured sparsity
models to perform room modeling and speech recovery. We propose a scheme for
characterizing the room acoustic from the unknown competing speech sources
relying on localization of the early images of the speakers by sparse
approximation of the spatial spectra of the virtual sources in a free-space
model. The images are then clustered exploiting the low-rank structure of the
spectro-temporal components belonging to each source. This enables us to
identify the early support of the room impulse response function and its unique
map to the room geometry. To further tackle the ambiguity of the reflection
ratios, we propose a novel formulation of the reverberation model and estimate
the absorption coefficients through a convex optimization exploiting joint
sparsity model formulated upon spatio-spectral sparsity of concurrent speech
representation. The acoustic parameters are then incorporated for separating
individual speech signals through either structured sparse recovery or inverse
filtering the acoustic channels. The experiments conducted on real data
recordings demonstrate the effectiveness of the proposed approach for
multi-party speech recovery and recognition.Comment: 31 page
Permutation Invariant Training of Deep Models for Speaker-Independent Multi-talker Speech Separation
We propose a novel deep learning model, which supports permutation invariant
training (PIT), for speaker independent multi-talker speech separation,
commonly known as the cocktail-party problem. Different from most of the prior
arts that treat speech separation as a multi-class regression problem and the
deep clustering technique that considers it a segmentation (or clustering)
problem, our model optimizes for the separation regression error, ignoring the
order of mixing sources. This strategy cleverly solves the long-lasting label
permutation problem that has prevented progress on deep learning based
techniques for speech separation. Experiments on the equal-energy mixing setup
of a Danish corpus confirms the effectiveness of PIT. We believe improvements
built upon PIT can eventually solve the cocktail-party problem and enable
real-world adoption of, e.g., automatic meeting transcription and multi-party
human-computer interaction, where overlapping speech is common.Comment: 5 page
Deep clustering: Discriminative embeddings for segmentation and separation
We address the problem of acoustic source separation in a deep learning
framework we call "deep clustering." Rather than directly estimating signals or
masking functions, we train a deep network to produce spectrogram embeddings
that are discriminative for partition labels given in training data. Previous
deep network approaches provide great advantages in terms of learning power and
speed, but previously it has been unclear how to use them to separate signals
in a class-independent way. In contrast, spectral clustering approaches are
flexible with respect to the classes and number of items to be segmented, but
it has been unclear how to leverage the learning power and speed of deep
networks. To obtain the best of both worlds, we use an objective function that
to train embeddings that yield a low-rank approximation to an ideal pairwise
affinity matrix, in a class-independent way. This avoids the high cost of
spectral factorization and instead produces compact clusters that are amenable
to simple clustering methods. The segmentations are therefore implicitly
encoded in the embeddings, and can be "decoded" by clustering. Preliminary
experiments show that the proposed method can separate speech: when trained on
spectrogram features containing mixtures of two speakers, and tested on
mixtures of a held-out set of speakers, it can infer masking functions that
improve signal quality by around 6dB. We show that the model can generalize to
three-speaker mixtures despite training only on two-speaker mixtures. The
framework can be used without class labels, and therefore has the potential to
be trained on a diverse set of sound types, and to generalize to novel sources.
We hope that future work will lead to segmentation of arbitrary sounds, with
extensions to microphone array methods as well as image segmentation and other
domains.Comment: Originally submitted on June 5, 201
LibriMix: An Open-Source Dataset for Generalizable Speech Separation
In recent years, wsj0-2mix has become the reference dataset for
single-channel speech separation. Most deep learning-based speech separation
models today are benchmarked on it. However, recent studies have shown
important performance drops when models trained on wsj0-2mix are evaluated on
other, similar datasets. To address this generalization issue, we created
LibriMix, an open-source alternative to wsj0-2mix, and to its noisy extension,
WHAM!. Based on LibriSpeech, LibriMix consists of two- or three-speaker
mixtures combined with ambient noise samples from WHAM!. Using Conv-TasNet, we
achieve competitive performance on all LibriMix versions. In order to fairly
evaluate across datasets, we introduce a third test set based on VCTK for
speech and WHAM! for noise. Our experiments show that the generalization error
is smaller for models trained with LibriMix than with WHAM!, in both clean and
noisy conditions. Aiming towards evaluation in more realistic,
conversation-like scenarios, we also release a sparsely overlapping version of
LibriMix's test set.Comment: submitted to INTERSPEECH 202
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