TasNet: time-domain audio separation network for real-time, single-channel speech separation

Robust speech processing in multi-talker environments requires effective speech separation. Recent deep learning systems have made significant progress toward solving this problem, yet it remains challenging particularly in real-time, short latency applications. Most methods attempt to construct a mask for each source in time-frequency representation of the mixture signal which is not necessarily an optimal representation for speech separation. In addition, time-frequency decomposition results in inherent problems such as phase/magnitude decoupling and long time window which is required to achieve sufficient frequency resolution. We propose Time-domain Audio Separation Network (TasNet) to overcome these limitations. We directly model the signal in the time-domain using an encoder-decoder framework and perform the source separation on nonnegative encoder outputs. This method removes the frequency decomposition step and reduces the separation problem to estimation of source masks on encoder outputs which is then synthesized by the decoder. Our system outperforms the current state-of-the-art causal and noncausal speech separation algorithms, reduces the computational cost of speech separation, and significantly reduces the minimum required latency of the output. This makes TasNet suitable for applications where low-power, real-time implementation is desirable such as in hearable and telecommunication devices.Comment: Camera ready version for ICASSP 2018, Calgary, Canad

Online Speaker Separation Using Deep Clustering

In this thesis, a low-latency variant of speaker-independent deep clustering method is proposed for speaker separation. Compared to the offline deep clustering separation system, bidirectional long-short term memory networks (BLSTMs) are replaced with long-short term memory networks (LSTMs). The reason is that the data has to be fed to the BLSTM networks both forward and backward directions. Additionally, the final outputs depend on both directions, which make online processing not possible. Also, 32 ms synthesis window is replaced with 8 ms in order to cooperate with low- latency applications like hearing aids since the algorithmic latency depends upon the length of synthesis window. Furthermore, the beginning of the audio mixture, here, referred as buffer, is used to get the cluster centers for the constituent speakers in the mixture serving as the initialization purpose. Later, those centers are used to assign clusters for the rest of the mixture to achieve speaker separation with the latency of 8 ms. The algorithm is evaluated on the Wall Street Journal corpus (WSJ0). Changing the networks from BLSTM to LSTM while keeping the same window length degrades the separation performance measured by signal-to-distortion ratio (SDR) by 1.0 dB, which implies that the future information is important for the separation. For investigating the effect of window length, keeping the same network structure (LSTM), by changing window length from 32 ms to 8 ms, another 1.1 dB drop in SDR is found. For the low-latency deep clustering speaker separation system, different duration of buffer is studied. It is observed that initially, the separation performance increases as the buffer increases. However, with buffer length of 0.3 s, the separation performance keeps steady even by increasing the buffer. Compared to offline deep clustering separation system, degradation of 2.8 dB in SDR is observed for online system

Online Monaural Speech Enhancement Using Delayed Subband LSTM

This paper proposes a delayed subband LSTM network for online monaural (single-channel) speech enhancement. The proposed method is developed in the short time Fourier transform (STFT) domain. Online processing requires frame-by-frame signal reception and processing. A paramount feature of the proposed method is that the same LSTM is used across frequencies, which drastically reduces the number of network parameters, the amount of training data and the computational burden. Training is performed in a subband manner: the input consists of one frequency, together with a few context frequencies. The network learns a speech-to-noise discriminative function relying on the signal stationarity and on the local spectral pattern, based on which it predicts a clean-speech mask at each frequency. To exploit future information, i.e. look-ahead, we propose an output-delayed subband architecture, which allows the unidirectional forward network to process a few future frames in addition to the current frame. We leverage the proposed method to participate to the DNS real-time speech enhancement challenge. Experiments with the DNS dataset show that the proposed method achieves better performance-measuring scores than the DNS baseline method, which learns the full-band spectra using a gated recurrent unit network.Comment: Paper submitted to Interspeech 202