In this thesis, we propose an artificial auditory system that gives a robot the ability to locate and track sounds, as well as to separate simultaneous sound sources and recognising simultaneous speech. We demonstrate that it is possible to implement these capabilities using an array of microphones, without trying to imitate the human auditory system. The sound source localisation and tracking algorithm uses a steered beamformer to locate sources, which are then tracked using a multi-source particle filter. Separation of simultaneous sound sources is achieved using a variant of the Geometric Source Separation (GSS) algorithm, combined with a multi-source post-filter that further reduces noise, interference and reverberation. Speech recognition is performed on separated sources, either directly or by using Missing Feature Theory (MFT) to estimate the reliability of the speech features. The results obtained show that it is possible to track up to four simultaneous sound sources, even in noisy and reverberant environments. Real-time control of the robot following a sound source is also demonstrated. The sound source separation approach we propose is able to achieve a 13.7 dB improvement in signal-to-noise ratio compared to a single microphone when three speakers are present. In these conditions, the system demonstrates more than 80 accuracy on digit recognition, higher than most human listeners could obtain in our small case study when recognising only one of these sources. All these new capabilities will allow humans to interact more naturally with a mobile robot in real life settings.
Research of confidence islands in the time-frequency plane representation has been shown to be effective in various applications and can be implemented with different strategies. One of the most effective is the missing feature strategy. Cooke @cite_14 @cite_3 propose a probabilistic estimation of a mask in regions of the time-frequency plane where the information is not reliable. Then, after masking, the parameters for speech recognition are generated and can be used in conventional speech recognition systems. Using this method, it is possible to obtain a significant increase in recognition rates without any modelling of the noise @cite_108 . In this scheme, the mask is essentially based on the signal-to-interference ratio (SIR) and a probabilistic estimation of the mask is used.
@cite_14 @cite_3 @cite_108
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This paper proposes an end-to-end approach for single-channel speaker-independent multi-speaker speech separation, where time-frequency (T-F) masking, the short-time Fourier transform (STFT), and its inverse are represented as layers within a deep network. Previous approaches, rather than computing a loss on the reconstructed signal, used a surrogate loss based on the target STFT magnitudes. This ignores reconstruction error introduced by phase inconsistency. In our approach, the loss function is directly defined on the reconstructed signals, which are optimized for best separation. In addition, we train through unfolded iterations of a phase reconstruction algorithm, represented as a series of STFT and inverse STFT layers. While mask values are typically limited to lie between zero and one for approaches using the mixture phase for reconstruction, this limitation is less relevant if the estimated magnitudes are to be used together with phase reconstruction. We thus propose several novel activation functions for the output layer of the T-F masking, to allow mask values beyond one. On the publicly-available wsj0-2mix dataset, our approach achieves state-of-the-art 12.6 dB scale-invariant signal-to-distortion ratio (SI-SDR) and 13.1 dB SDR, revealing new possibilities for deep learning based phase reconstruction and representing a fundamental progress towards solving the notoriously-hard cocktail party problem. Recently, time-frequency mask-based beamforming has been extensively studied as the frontend of deep neural network (DNN) based automatic speech recognition (ASR) in noisy environments. Two mask estimation approaches have been separately developed for this beamforming method, namely the the DNN-based approach, which exploits the time-frequency features of the signal, and the spatial clustering-based approach, which exploits the spatial features of the signal. This paper proposes a new method that integrates the two approaches in a probabilistic way to further improve mask estimation by exploiting the advantages of both approaches. Experiments using the real data of the CHiME-3 multichannel noisy speech corpus show that the proposed method almost always outperforms the conventional approaches in terms of word error rate (WER) improvement. Abstract Human speech perception is robust in the face of a wide variety of distortions, both experimentally applied and naturally occurring. In these conditions, state-of-the-art automatic speech recognition (ASR) technology fails. This paper describes an approach to robust ASR which acknowledges the fact that some spectro-temporal regions will be dominated by noise. For the purposes of recognition, these regions are treated as missing or unreliable. The primary advantage of this viewpoint is that it makes minimal assumptions about any noise background. Instead, reliable regions are identified, and subsequent decoding is based on this evidence. We introduce two approaches for dealing with unreliable evidence. The first – marginalisation – computes output probabilities on the basis of the reliable evidence only. The second – state-based data imputation – estimates values for the unreliable regions by conditioning on the reliable parts and the recognition hypothesis. A further source of information is the bounds on the energy of any constituent acoustic source in an additive mixture. This additional knowledge can be incorporated into the missing data framework. These approaches are applied to continuous-density hidden Markov model (HMM)-based speech recognisers and evaluated on the TIDigits corpus for several noise conditions. Two criteria which use simple noise estimates are employed as a means of identifying reliable regions. The first treats regions which are negative after spectral subtraction as unreliable. The second uses the estimated noise spectrum to derive local signal-to-noise ratios, which are then thresholded to identify reliable data points. Both marginalisation and state-based data imputation produce a substantial performance advantage over spectral subtraction alone. The use of energy bounds leads to a further increase in performance for both approaches. While marginalisation outperforms data imputation, the latter technique allows the technique to act as a preprocessor for conventional recognisers, or in speech-enhancement applications. This paper considers acoustic beamforming for noise robust automatic speech recognition. A beamformer attenuates background noise by enhancing sound components coming from a direction specified by a steering vector. Hence, accurate steering vector estimation is paramount for successful noise reduction. Recently, time-frequency masking has been proposed to estimate the steering vectors that are used for a beamformer. In particular, we have developed a new form of this approach, which uses a speech spectral model based on a complex Gaussian mixture model CGMM to estimate the time-frequency masks needed for steering vector estimation, and extended the CGMM-based beamformer to an online speech enhancement scenario. Our previous experiments showed that the proposed CGMM-based approach outperforms a recently proposed mask estimator based on a Watson mixture model and the baseline speech enhancement system of the CHiME-3 challenge. This paper provides additional experimental results for our online processing, which achieves performance comparable to that of batch processing with a suitable block-batch size. This online version reduces the CHiME-3 word error rate WER on the evaluation set from 8.37 to 8.06 . Moreover, in this paper, we introduce a probabilistic prior distribution for a spatial correlation matrix a CGMM parameter, which enables more stable steering vector estimation in the presence of interfering speakers. In practice, the performance of the proposed online beamformer degrades with observations that contain only noise or and interference because of the failure of the CGMM parameter estimation. The introduced spatial prior enables the target speaker's parameter to avoid overfitting to noise or and interference. Experimental results show that the spatial prior reduces the WER from 38.4 to 29.2 in a conversation recognition task compared with the CGMM-based approach without the prior, and outperforms a conventional online speech enhancement approach.