There is, I think, something ethereal about i —the square root of minus one. I remember first hearing about it at school. It seemed an odd beast at that time—an intruder hovering on the edge of reality.

Usually familiarity dulls this sense of the bizarre, but in the case of i it was the reverse: over the years the sense of its surreal nature intensified. It seemed that it was impossible to write mathematics that described the real world in …

I and i

Single-channel, speaker-independent speech separation methods have recently seen great progress. However, the accuracy, latency, and computational cost of such methods remain insufficient. The majority of the previous methods have formulated the separation problem through the time-frequency representation of the mixed signal, which has several drawbacks, including the decoupling of the phase and magnitude of the signal, the suboptimality of time-frequency representation for speech separation, and the long latency in calculating the spectrograms. To address these shortcomings, we propose a fully-convolutional time-domain audio separation network (Conv-TasNet), a deep learning framework for end-to-end time-domain speech separation. Conv-TasNet uses a linear encoder to generate a representation of the speech waveform optimized for separating individual speakers. Speaker separation is achieved by applying a set of weighting functions (masks) to the encoder output. The modified encoder representations are then inverted back to the waveforms using a linear decoder. The masks are found using a temporal convolutional network (TCN) consisting of stacked 1-D dilated convolutional blocks, which allows the network to model the long-term dependencies of the speech signal while maintaining a small model size. The proposed Conv-TasNet system significantly outperforms previous time-frequency masking methods in separating two- and three-speaker mixtures. Additionally, Conv-TasNet surpasses several ideal time-frequency magnitude masks in two-speaker speech separation as evaluated by both objective distortion measures and subjective quality assessment by human listeners. Finally, Conv-TasNet has a significantly smaller model size and a shorter minimum latency, making it a suitable solution for both offline and real-time speech separation applications.

/pdf/conv-tasnet-surpassing-ideal-time-frequency-magnitude-3fbpvuuk37.pdf

Conv-TasNet: Surpassing Ideal Time-Frequency Magnitude Masking for Speech Separation

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Detection, Estimation, and Modulation Theory

Speech enhancement and separation are core problems in audio signal processing, with commercial applications in devices as diverse as mobile phones, conference call systems, hands-free systems, or hearing aids. In addition, they are crucial preprocessing steps for noise-robust automatic speech and speaker recognition. Many devices now have two to eight microphones. The enhancement and separation capabilities offered by these multichannel interfaces are usually greater than those of single-channel interfaces. Research in speech enhancement and separation has followed two convergent paths, starting with microphone array processing and blind source separation, respectively. These communities are now strongly interrelated and routinely borrow ideas from each other. Yet, a comprehensive overview of the common foundations and the differences between these approaches is lacking at present. In this paper, we propose to fill this gap by analyzing a large number of established and recent techniques according to four transverse axes: 1 the acoustic impulse response model, 2 the spatial filter design criterion, 3 the parameter estimation algorithm, and 4 optional postfiltering. We conclude this overview paper by providing a list of software and data resources and by discussing perspectives and future trends in the field.

/pdf/a-consolidated-perspective-on-multimicrophone-speech-qnbaqxmtvq.pdf

A Consolidated Perspective on Multimicrophone Speech Enhancement and Source Separation

2017 25th European Signal Processing Conference (EUSIPCO)

Microphone array processing utilize spatial separation between the desired speaker and interference signal for speech enhancement. The transfer functions (TFs) relating the speaker component at a reference microphone with all other microphones, denoted as the relative TFs (RTFs), play an important role in beamforming design criteria such as minimum variance distortionless response (MVDR) and speech distortion weighted multichannel Wiener filter (SDW-MWF). Two common methods for estimating the RTF are surveyed here, namely, the covariance subtraction (CS) and the covariance whitening (CW) methods. We analyze the performance of the CS method theoretically and empirically validate the results of the analysis through extensive simulations. Furthermore, empirically comparing the methods performances in various scenarios evidently shows thats the CW method outperforms the CS method.

Performance analysis of the covariance subtraction method for relative transfer function estimation and comparison to the covariance whitening method

http://www.eng.biu.ac.il/~gannot/articles/Optimal%20distributed%20minimum-variance%20beamforming.pdf

Optimal distributed minimum-variance beamforming approaches for speech enhancement in wireless acoustic sensor networks

This paper proposes a distributed multiple constraints generalized sidelobe canceler (GSC) for speech enhancement in an N-node fully connected wireless acoustic sensor network (WASN) comprising M microphones. Our algorithm is designed to operate in reverberant environments with constrained speakers (including both desired and competing speakers). Rather than broadcasting M microphone signals, a significant communication bandwidth reduction is obtained by performing local beamforming at the nodes, and utilizing only transmission channels. Each node processes its own microphone signals together with the N + P transmitted signals. The GSC-form implementation, by separating the constraints and the minimization, enables the adaptation of the BF during speech-absent time segments, and relaxes the requirement of other distributed LCMV based algorithms to re-estimate the sources RTFs after each iteration. We provide a full convergence proof of the proposed structure to the centralized GSC-beamformer (BF). An extensive experimental study of both narrowband and (wideband) speech signals verifies the theoretical analysis.

Distributed Multiple Constraints Generalized Sidelobe Canceler for Fully Connected Wireless Acoustic Sensor Networks

Beamforming methods for speech enhancement in wireless acoustic sensor networks (WASNs) have recently attracted the attention of the research community. One of the major obstacles in implementing speech processing algorithms in WASN is the sampling rate offsets between the nodes. As nodes utilize individual clock sources, sampling rate offsets are inevitable and may cause severe performance degradation. In this paper, a blind procedure for estimating the sampling rate offsets is derived. The procedure is applicable to speech-absent time segments with slow time-varying interference statistics. The proposed procedure is based on the phase drift of the coherence between two signals sampled at different sampling rates. Resampling the signals with Lagrange polynomials interpolation method compensates for the sampling rate offsets. An extensive experimental study, utilizing the transfer function generalized sidelobe canceller (TFGSC), exemplifies the problem and its solution.

/pdf/blind-sampling-rate-offset-estimation-and-compensation-in-2jnuz9mb8c.pdf

Shmulik Markovich-Golan

Papers

A Consolidated Perspective on Multimicrophone Speech Enhancement and Source Separation

Performance analysis of the covariance subtraction method for relative transfer function estimation and comparison to the covariance whitening method

Optimal distributed minimum-variance beamforming approaches for speech enhancement in wireless acoustic sensor networks

Distributed Multiple Constraints Generalized Sidelobe Canceler for Fully Connected Wireless Acoustic Sensor Networks

Blind Sampling Rate Offset Estimation and Compensation in Wireless Acoustic Sensor Networks with Application to Beamforming