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

Non-Negative Matrix Factorization Based Algorithms to Cluster Frequency Basis Functions for Monaural Sound Source Separation.

Monophonic sound source separation (SSS) refers to a process that separates out audio signals produced from the individual sound sources in a given acoustic mixture, when the mixture signal is recorded using one microphone or is directly recorded onto one reproduction channel. Many audio applications such as pitch modification and automatic music transcription would benefit from the availability of segregated sound sources from the mixture of audio signals for further processing. Recently, Non-negative matrix factorization (NMF) has found application in monaural audio source separation due to its ability to factorize audio spectrograms into additive part-based basis functions, where the parts typically correspond to individual notes or chords in music. An advantage of NMF is that there can be a single basis function for each note played by a given instrument, thereby capturing changes in timbre with pitch for each instrument or source. However, these basis functions need to be clustered to their respective sources for the reconstruction of the individual source signals. Many clustering methods have been proposed to map the separated signals into sources with considerable success. Recently, to avoid the need of clustering, Shifted NMF (SNMF) was proposed, which assumes that the timbre of a note is constant for all the pitches produced by an instrument. SNMF has two drawbacks. Firstly, the assumption that the timbre of the notes played by an instrument remains constant, is not true in general. Secondly, the SNMF method uses the Constant Q transform (CQT) and the lack of a true inverse of the CQT results in compromising on separation quality of the reconstructed signal. The principal aim of this thesis is to attempt to solve the problem of clustering NMF basis functions. Our first major contribution is the use of SNMF as a method of clustering the basis functions obtained via standard NMF. The proposed SNMF clustering method aims to cluster the frequency basis functions obtained via standard NMF to their respective sources by making use of shift invariance in a log-frequency domain. Further, a minor contribution is made by improving the separation performance of the standard SNMF algorithm (here used directly to separate sources) obtained through the use of an improved inverse CQT. Here, the standard SNMF algorithm finds shift-invariance in a CQ spectrogram, that contain the frequency basis functions, obtained directly from the spectrogram of the audio mixture. Our next contribution is an improvement in the SNMF clustering algorithm through the incorporation of the CQT matrix inside the SNMF model in order to avoid the need of an inverse CQT to reconstruct the clustered NMF basis unctions. Another major contribution deals with the incorporation of a constraint called group sparsity (GS) into the SNMF clustering algorithm at two stages to improve clustering. The effect of the GS is evaluated on various SNMF clustering algorithms proposed in this thesis. Finally, we have introduced a new family of masks to reconstruct the original signal from the clustered basis functions and compared their performance to the generalized Wiener filter masks using three different factorisation-based separation algorithms. We show that better separation performance can be achieved by using the proposed family of masks

Multi-talker Speech Separation with Utterance-level Permutation Invariant Training of Deep Recurrent Neural Networks

In this paper we propose the utterance-level Permutation Invariant Training (uPIT) technique. uPIT is a practically applicable, end-to-end, deep learning based solution for speaker independent multi-talker speech separation. Specifically, uPIT extends the recently proposed Permutation Invariant Training (PIT) technique with an utterance-level cost function, hence eliminating the need for solving an additional permutation problem during inference, which is otherwise required by frame-level PIT. We achieve this using Recurrent Neural Networks (RNNs) that, during training, minimize the utterance-level separation error, hence forcing separated frames belonging to the same speaker to be aligned to the same output stream. In practice, this allows RNNs, trained with uPIT, to separate multi-talker mixed speech without any prior knowledge of signal duration, number of speakers, speaker identity or gender. We evaluated uPIT on the WSJ0 and Danish two- and three-talker mixed-speech separation tasks and found that uPIT outperforms techniques based on Non-negative Matrix Factorization (NMF) and Computational Auditory Scene Analysis (CASA), and compares favorably with Deep Clustering (DPCL) and the Deep Attractor Network (DANet). Furthermore, we found that models trained with uPIT generalize well to unseen speakers and languages. Finally, we found that a single model, trained with uPIT, can handle both two-speaker, and three-speaker speech mixtures