Denoising sound signals in a bioinspired non-negative spectro-temporal domain

The representation of sound signals at the cochlea and auditory cortical level has been studied as an alternative to classical analysis methods. In this work, we put forward a recently proposed feature extraction method called approximate auditory cortical representation, based on an approximation to the statistics of discharge patterns at the primary auditory cortex. The approach here proposed estimates a non-negative sparse coding with a combined dictionary of atoms. These atoms represent the spectro-temporal receptive fields of the auditory cortical neurons, and are calculated from the auditory spectrograms of clean signal and noise. The denoising is carried out on noisy signals by the reconstruction of the signal discarding the atoms corresponding to the noise. Experiments are presented using synthetic (chirps) and real data (speech), in the presence of additive noise. For the evaluation of the new method and its variants, we used two objective measures: the perceptual evaluation of speech quality and the segmental signal-to-noise ratio. Results show that the proposed method improves the quality of the signals, mainly under severe degradation.     

The Sensitivity Matrix: Using Advanced Auditory Models in Speech and Audio Processing

Perceptually optimal processing of speech and audio signals demands a rigorous approach using a distortion measure that resembles human perception. This requires distortion measures based on sophisticated, complex auditory models. Under the assumption of small distortions these models can be simplified by means of a sensitivity matrix. In this paper, we show the power of this approach. We present a method to derive the sensitivity matrix for distortion measures based on spectro-temporal auditory models. This method is applied to an example auditory model and the region of validity of the approximation and the application of linear algebra to analyze the characteristics of the given model are discussed. Furthermore, we show how to build a coder minimizing a sensitivity matrix distortion measure given the typically long support of a perceptual distortion measure     

Short-term synaptic plasticity can enhance weak signal detectability in nonrenewal spike trains

We study the encoding of weak signals in spike trains with interspike interval (ISI) correlations and the signals' subsequent detection in sensory neurons. Motivated by the observation of negative ISI correlations in auditory and electrosensory afferents, we assess the theoretical performance limits of an individual detector neuron receiving a weak signal distributed across multiple afferent inputs. We assess the functional role of ISI correlations in the detection process using statistical detection theory and derive two sequential likelihood ratio detector models: one for afferents with renewal statistics; the other for afferents with negatively correlated ISIs. We suggest a mechanism that might enable sensory neurons to implicitly compute conditional probabilities of presynaptic spikes by means of short-term synaptic plasticity. We demonstrate how this mechanism can enhance a postsynaptic neuron's sensitivity to weak signals by exploiting the correlation structure of the input spike trains. Our model not only captures fundamental aspects of early electrosensory signal processing in weakly electric fish, but may also bear relevance to the mammalian auditory system and other sensory modalities.     

基于新型混合模型的欠定盲分离方法

针对欠定盲分离问题,提出了一种新的源恢复方法. 在时频域局部区域采用复高斯分布对源信号进行建模,将语音信号的稀疏性和局部平稳性结合在一起,提出了一种新的混合模型来描述观测信号在局部区域的概率分布.通过该模型,将每个时频点的源信号状态的判断问题转换成模型的参数估计和后验概率的计算问题,最后通过子混合矩阵的逆恢复出源信号. 实验结果表明,该方法具有很快的收敛速度,并且比已有方法具有更好的分离性能.     

Efficient coding of time-relative structure using spikes

Nonstationary acoustic features provide essential cues for many auditory tasks, including sound localization, auditory stream analysis, and speech recognition. These features can best be characterized relative to a precise point in time, such as the onset of a sound or the beginning of a harmonic periodicity. Extracting these types of features is a difficult problem. Part of the difficulty is that with standard block-based signal analysis methods, the representation is sensitive to the arbitrary alignment of the blocks with respect to the signal. Convolutional techniques such as shift-invariant transformations can reduce this sensitivity, but these do not yield a code that is efficient, that is, one that forms a nonredundant representation of the underlying structure. Here, we develop a non-block-based method for signal representation that is both time relative and efficient. Signals are represented using a linear superposition of time-shiftable kernel functions, each with an associated magnitude and temporal position. Signal decomposition in this method is a non-linear process that consists of optimizing the kernel function scaling coefficients and temporal positions to form an efficient, shift-invariant representation. We demonstrate the properties of this representation for the purpose of characterizing structure in various types of nonstationary acoustic signals. The computational problem investigated here has direct relevance to the neural coding at the auditory nerve and the more general issue of how to encode complex, time-varying signals with a population of spiking neurons.     

Blind source separation based on constant modulus criterion and signal mutual information

We address the problem of adaptive blind source separation (BSS) from instantaneous multi-input multi-output (MIMO) channels. It is known that the constant modulus (CM) criterion can be used to extract unknown source signals. However, the existing CM-based algorithms normally extract the source signals in a serial manner. Consequently, the accuracy in extracting each source signal, except for the first one, depends on the accuracy of previous source extraction. This estimation error propagation (accumulation) will cause severe performance degradation. In this letter, we propose a new adaptive separation algorithm that can separate all source signals simultaneously by directly updating the separation matrix. The superior performance of the new algorithm is demonstrated by simulation examples.     

Three-dimensional auditory display for enhancing detection of passive sonar signals

OBJECTIVE: The viability of a three-dimensional (3-D) auditory display for improving signal detection of passive sonar signals was investigated. BACKGROUND: Sonar operators usually have difficulty detecting targets because the sound received by the hydrophone has a low signal-to-noise ratio when coupled with the operator's headset that does not isolate well against the ambient noise. METHODS: Release from masking was assessed by pairing a recording of a torpedo with diotic broadband pink noise that served as a masker, and a 400 Hz tone with the masker. Masked thresholds were measured for seven signal durations when each signal was presented dioticly and in 3-D auditory space at three positions on the horizontal plane. RESULTS: The spatial separation of signal and masker yielded a significant improvement in detection. CONCLUSION: A 3-D auditory display is a viable technology that could lead to a significant improvement in release from masking. The magnitude of the masking level difference will vary with respect to the characteristics of the hydrophone signal and masker and the synthesis capability of the 3-D auditory display. APPLICATION: Potential applications of this research include enhanced auditory displays for processing passive sonar signals, leading to earlier detection of enemy targets.     

Analysis and synthesis of speech using least <Emphasis Type="Italic">P</Emphasis>th norm filter design

In this paper we analyze the combination of speech and FIR filter design aspect to achieve good results in speech quality. A new approach in the time domain based on the least Pth norm is presented to extract maximum information that represents speech. The aim of this paper is to improve the perceived quality of speech through the introduction of least Pth norm algorithm that attenuates speech contaminated with noise. This approach relates to a filter bank structure and a method for filtering and separating an information signal into different bands, particularly for filtering and separation of speech signals. Then the desired signal is reconstructed from the independent components representing every band. This approach differs from the traditional approaches since no priori knowledge of the noise statistics is required, instead the noise signals are only assumed to have finite energy. Since the estimation criterion for the filter design is to minimize the worst possible amplification of the estimation error signal in terms of modeling errors and additive noise, this approach is highly robust and appropriate in practical speech analysis and synthesis. This paper presents a least Pth approach to the optimal design of FIR digital filter banks in the minimax sense for speech analysis and synthesis. The signal to noise ratio (SNR) of around 50–60 dB is achieved with various speech samples.     

基于广义解调时频分析的多分量复杂信号分解方法

在广义解调时频分析方法的基础上,将该方法应用于多分量时频变化信号的分解,并对该方法进行了改进,给出了由改进的广义解调时频分析方法分解多分量复杂信号的具体步骤.重点分析了自动获取相位函数的方法,以及对分离出来的各单分量再次进行广义时频解调的问题,得到了比较理想的时频分布.仿真实验结果证明:此法不仅可以得到原始信号中各分量的时域波形,同时也可以得到各分量的时频分布,从而为多分量复杂时频变化信号的准确分离提供了有效途径.     

Independent component analysis based on nonparametric density estimation

In this paper, we introduce a novel independent component analysis (ICA) algorithm, which is truly blind to the particular underlying distribution of the mixed signals. Using a nonparametric kernel density estimation technique, the algorithm performs simultaneously the estimation of the unknown probability density functions of the source signals and the estimation of the unmixing matrix. Following the proposed approach, the blind signal separation framework can be posed as a nonlinear optimization problem, where a closed form expression of the cost function is available, and only the elements of the unmixing matrix appear as unknowns. We conducted a series of Monte Carlo simulations, involving linear mixtures of various source signals with different statistical characteristics and sample sizes. The new algorithm not only consistently outperformed all state-of-the-art ICA methods, but also demonstrated the following properties: 1) Only a flexible model, capable of learning the source statistics, can consistently achieve an accurate separation of all the mixed signals. 2) Adopting a suitably designed optimization framework, it is possible to derive a flexible ICA algorithm that matches the stability and convergence properties of conventional algorithms. 3) A nonparametric approach does not necessarily require large sample sizes in order to outperform methods with fixed or partially adaptive contrast functions.     

智能化高精度脉宽数字滤波电路的设计

在信号的传输过程中,噪音信号不可避免地会干扰正常信号.如何对混合信号进行信噪分离,是目前业界普遍关注和研究的问题.针对高频方波信号的信噪分离问题,提出了智能化脉宽数字滤波的技术方案,并设计了一种实用的脉宽数字滤波电路,消除了测量信号中的尖峰干扰成分.实践证明：该电路能有效地实现高频方波信号与尖峰脉冲干扰信号的信噪分离.     

Morphometrical data analysis using wavelets

In this paper, we present a new shape analysis approach using the well-known wavelet transform and exploring shape representation by landmarks. First, we describe the approach adopted to represent the landmarks data as parametric signals. Then, we show the relation of the derivatives of Gaussian wavelet transform applied to the signal-to-differential properties of the shape that it represents. We present experimental results using real data to show how it is possible to characterize shapes through multiscale and differential signal-processing techniques in order to relate morphological variables with phylogenetic signal, environmental factors and sexual dimorphism. The goal of this research is to develop an effective wavelet transform-based method to represent and classify multiple classes of shapes given by landmarks.     

Blind source separation by nonstationarity of variance: acumulant-based approach

Blind separation of source signals usually relies either on the nonGaussianity of the signals or on their linear autocorrelations. A third approach was introduced by Matsuoka et al. (1995), who showed that source separation can be performed by using the nonstationarity of the signals, in particular the nonstationarity of their variances. In this paper, we show how to interpret the nonstationarity due to a smoothly changing variance in terms of higher order cross-cumulants. This is based on the time-correlation of the squares (energies) of the signals and leads to a simple optimization criterion. Using this criterion, we construct a fixed-point algorithm that is computationally very efficient.     

An unsupervised approach for co-channel speech separation using Hilbert–Huang transform and Fuzzy C-Means clustering

In this paper we discuss an unsupervised approach for co-channel speech separation where two speakers are speaking simultaneously over same channel. We propose a two stage separation process where the initial stage is based on empirical mode decomposition (EMD) and Hilbert transform generally known as Hilbert–Huang transform. EMD decomposes the mixed signal into oscillatory functions known as intrinsic mode functions. Hilbert transform is applied to find the instantaneous amplitudes and Fuzzy C-Means clustering is applied to group the speakers at initial stage. In second stage of separation speaker groups are transformed into time–frequency domain using short time Fourier transform (STFT). Time–frequency ratio’s are computed by dividing the STFT matrix of mixed speech signal and STFT matrix of stage1 recovered speech signals. Histogram of the ratios obtained can be used to estimate the ideal binary mask for each speaker. These masks are applied to the speech mixture and the underlying speakers are estimated. Masks are estimated from the speech mixture and helps in imputing the missing values after stage1 grouping of speakers. Results obtained show significant improvement in objective measures over other existing single-channel speech separation methods.     

基于特征值分解的循环平稳离散时间信号的盲分离

盲分离的目的是从观测到的混叠信号中恢复出各个未知的源信号,现今的很多方法都是利用了信号时域表示的某些统计特性来解决这个问题。从信号频域分析的角度提出了一种利用信号的循环平稳特性来处理离散时间信号的频域盲分离方法。该方法构造两个二阶统计矩阵的乘积,并对该乘积矩阵进行特征值分解,从而实现源信号的分离;同时,还对特征值分解的条件进行了分析。该方法在低维信号的情况下可以取得相当满意的分离效果,仿真结果表明该方法具有良好的性能。     

Pattern recognition approach to identify natural clusters of acoustic emission signals

A new approach is introduced to identify natural clusters of acoustic emission signals. The presented technique is based on an exhaustive screening taking into account all combinations of signal features extracted from the recorded acoustic emission signals. For each possible combination of signal features an investigation of the classification performance of the k-means algorithm is evaluated ranging from two to ten classes. The numerical degree of cluster separation of each partition is calculated utilizing the Davies-Bouldin and Tou indices, Rousseeuw’s silhouette validation method and Hubert’s Gamma statistics. The individual rating of each cluster validation technique is cumulated based on a voting scheme and is evaluated for the number of clusters with best performance. This is defined as the best partitioning for the given signal feature combination. As a second step the numerical ranking of all these partitions is evaluated for the globally optimal partition in a second voting scheme using the cluster validation methods results. This methodology can be used as an automated evaluation of the number of natural clusters and their partitions without previous knowledge about the cluster structure of acoustic emission signals. The suitability of the current approach was evaluated using artificial datasets with defined degree of separation. In addition the application of the approach to clustering of acoustic emission signals is demonstrated for signals obtained from failure during loading of carbon fiber reinforced plastic specimens.     

Signal Transduction Pathways as Concurrent Reactive Systems: A Modeling and Simulation Approach Using LSCs and the Play-Engine

Signal transduction pathways describe how cells respond to extracellular signals which are received by receptors at the cell membrane and usually transferred into the nucleus. In this paper we present our approach to model these signal transduction pathways as concurrent reactive systems by means of Life Sequence Charts and to simulate them using the Play-Engine tool. This aspect is part of a bigger approach, where we provide an extendable system to generate models of signal transduction pathways in different modeling languages and to simulate these models with the corresponding simulation tools.     

Efficient blind dereverberation and echo cancellation based on independent component analysis for actual acoustic signals

This letter presents a new algorithm for blind dereverberation and echo cancellation based on independent component analysis (ICA) for actual acoustic signals. We focus on frequency domain ICA (FD-ICA) because its computational cost and speed of learning convergence are sufficiently reasonable for practical applications such as hands-free speech recognition. In applying conventional FD-ICA as a preprocessing of automatic speech recognition in noisy environments, one of the most critical problems is how to cope with reverberations. To extract a clean signal from the reverberant observation, we model the separation process in the short-time Fourier transform domain and apply the multiple input/output inverse-filtering theorem (MINT) to the FD-ICA separation model. A naive implementation of this method is computationally expensive, because its time complexity is the second order of reverberation time. Therefore, the main issue in dereverberation is to reduce the high computational cost of ICA. In this letter, we reduce the computational complexity to the linear order of the reverberation time by using two techniques: (1) a separation model based on the independence of delayed observed signals with MINT and (2) spatial sphering for preprocessing. Experiments show that the computational cost grows in proportion to the linear order of the reverberation time and that our method improves the word correctness of automatic speech recognition by 10 to 20 points in a RT??= 670 ms reverberant environment.