基于子带广义旁瓣相消器的麦克风阵列语音增强*

为了加快基于广义旁瓣相消器的麦克风阵列语音增强系统的收敛速度,将其自适应模块的输入信号分解到子带以进行处理,并将多通道维纳滤波器引入广义旁瓣相消器的非自适应支路,以更有效地抑制非相干噪声。实际测试结果表明,相对于基于全带广义旁瓣相消器的麦克风阵列语音增强系统,采用该子带广义旁瓣相消器结构的语音增强系统具有更快的收敛速度和更高的输出信噪比。     

混响环境下的宽带波束形成语音增强方法

结合小波滤波器组理论和自适应波束形成技术,提出了一种基于宽带波束形成的麦克风阵列语音增强方法。该方法利用小波分析滤波器组将含噪语音信号变换到小波域;进行小波域阵列自适应波束形成;通过小波综合滤波器组重构增强后的语音信号。计算机仿真实验验证了该方法的有效性。     

语音信号共振峰频率估计的分段线性预测算法

基于分段线性预测算法估计语音的共振峰频率,运用多通道的滤波器组对语音的频段进行划分,然后选择合适的逆滤波器逼近不同频段的短时频谱,最后依据该逆滤波器估计共振峰频率。实验结果表明,与传统方法相比,该方法提高了语音共振峰频率估计时的分辨率与准确性,受噪声的影响较小。     

基于MATLAB的多通道人耳模型技术在皮肤听声中的应用

针对第一代皮肤听声器不能够准确进行语音辨析的问题,对人耳模型进行了多通道探索。根据中心频率和带宽设计了FIR滤波器,将300Hz～4kHz的音频范围内的音频信号划分为12个通道,音频信号经过12通道人耳模型的带通滤波,形成了电极片上的不同刺激;在MATLAB环境下实现了语音信号的12通道带通滤波及画出了相关语音的频谱图;仿真结果证明多通道人耳模型技术运用于皮肤听声系统中能够提高系统的稳定性,增强了辨析语音的能力。     

多通道皮肤听声系统中语音增强算法的应用研究

针对提高应用多通道皮肤听声系统进行语音识别的识别率,提出了基于多频带谱减法的语音增强算法。在多通道皮肤听声的实验中,有色噪声会严重降低语音质量,进而降低皮肤听声系统语音识别的识别率,因而首次将基于多带谱减法的语音增强算法引入到皮肤听声系统中以降低有色噪声。多频带谱减法将语音频带划分为多个子频带,分别在每个子频带作不同系数的谱减运算实现语音增强。通过Matlab完成了算法仿真并通过DSP硬件实现了算法并将增强后的语音信号输出给皮肤听声系统,实验证明此设计能够有效抑制有色噪声,增强皮肤听声系统的可靠性和实用性。     

一种低信噪比下的说话人识别算法研究

为了提高低信噪比下说话人识别系统的性能,提出一种Gammatone滤波器组与改进谱减法的语音增强相结合的说话人识别算法。将改进的谱减法作为预处理器,进一步提高语音信号的信噪比,再通过Gammatone滤波器组,对增强后的说话人语音信号进行处理,提取说话人语音信号的特征参数GFCC,进而将特征参数GFCC用于说话人识别算法中。仿真实验在高斯混合模型识别系统中进行。实验结果表明,采用这种算法应用于说话人识别系统,系统的识别率及鲁棒性都有明显的提高。     

结合PECGTFs和SSMC的腭裂语音咽擦音自动检测算法

咽擦音是腭裂语音中一种常见的代偿性构音异常,咽擦音的自动检测对腭咽功能的评估具有重要的临床意义。对腭裂语音咽擦音的自动检测算法进行了研究,提出分段指数压缩Gamatone滤波器组（Piecewise Exponent Compression Gammatone Filters,PECGTFs）和基于Softsign的多通道（Softsign-based Multi-Channel,SSMC）模型相结合提取语音信号的谱特征参数,采用KNN分类器,实现腭裂语音咽擦音的自动检测。实验共测试306个语音样本,并对比了使用不同的Gammatone滤波器、使用高斯差分（Difference of Gaussian,DoG）增强和SSMC模型增强对咽擦音自动检测结果的影响。实验结果表明,使用PECGTFs与SSMC相结合的算法对腭裂语音咽擦音的自动检测正确率达94.95%,对临床诊断具有一定的参考价值。     

基于CRV-LMS算法的语音增强技术的研究

在语音增强系统中,传统的LMS算法存在剩余均方误差随有用信号功率线性增加的问题。本文提出CRV-LMS算法,该算法首先设计组合滤波器,然后在滤波器中采用变步长系数,通过设计变步长更新律,从而使新权值更新收敛系数与输出功率成反比,达到提高语音增强系统中性能的目的。     

子空间与维纳滤波相结合的语音增强方法

针对复杂背景噪声下语音增强后带有音乐噪声的问题,提出一种子空间与维纳滤波相结合的语音增强方法。对带噪语音进行KL变换,估计出纯净语音的特征值,再利用子空间域中的信噪比计算公式构成一个维纳滤波器,使该特征值通过这个滤波器,从而得到新的纯净语音特征值,由KL逆变换还原出纯净语音。仿真结果表明,在白噪声和火车噪声的背景下,信噪比都比传统子空间方法有明显提高,并有效抑制了增强后产生的音乐噪声。     

GPS调零天线的数字接收机阵列

研究了GPS调零天线系统数字接收机阵列的一种高效实现方案.提出采用新型的可配置时分复用FIR滤波器组,实现了多通道数字下变频处理中的采样速率转换与滤波器系数重加载,通过对GPS调零天线系统四通道数字接收机阵列的设计实现与测试分析可以得出,此方法与多通道并行重复处理的常规数字下变频方式相比,可以节约大约60%的FPGA硬件资源,并可增强系统实现的灵活性.     

Speech enhancement based on undecimated wavelet packet-perceptual filterbanks and MMSE–STSA estimation in various noise environments

In this paper, we proposed a new speech enhancement system, which integrates a perceptual filterbank and minimum mean square error–short time spectral amplitude (MMSE–STSA) estimation, modified according to speech presence uncertainty. The perceptual filterbank was designed by adjusting undecimated wavelet packet decomposition (UWPD) tree, according to critical bands of psycho-acoustic model of human auditory system. The MMSE–STSA estimation (modified according to speech presence uncertainty) was used for estimation of speech in undecimated wavelet packet domain. The perceptual filterbank provides a good auditory representation (sufficient frequency resolution), good perceptual quality of speech and low computational load. The MMSE–STSA estimator is based on a priori SNR estimation. A priori SNR estimation, which is a key parameter in MMSE–STSA estimator, was performed by using “decision directed method.” The “decision directed method” provides a trade off between noise reduction and signal distortion when correctly tuned. The experiments were conducted for various noise types. The results of proposed method were compared with those of other popular methods, Wiener estimation and MMSE–log spectral amplitude (MMSE–LSA) estimation in frequency domain. To test the performance of the proposed speech enhancement system, three objective quality measurement tests (SNR, segSNR and Itakura–Saito distance (ISd)) were conducted for various noise types and SNRs. Experimental results and objective quality measurement test results proved the performance of proposed speech enhancement system. The proposed speech enhancement system provided sufficient noise reduction and good intelligibility and perceptual quality, without causing considerable signal distortion and musical background noise.     

Robust Speaker Identification and Verification

Acoustic characteristics have played an essential role in biometrics. In this article, we introduce a robust, text-independent speaker identification/verification system. This system is mainly based on a subspace-based enhancement technique and probabilistic support vector machines (SVMs). First, a perceptual filterbank is created from a psycho-acoustic model into which the subspace-based enhancement technique is incorporated. We use the prior SNR of each subband within the perceptual filterbank to decide the estimator's gain to effectively suppress environmental background noises. Then, probabilistic SVMs identify or verify the speaker from the enhanced speech. The superiority of the proposed system has been demonstrated by twenty speaker data taken from AURORA-2 database with added background noises     

Auditory inspired methods for localization of multiple concurrent speakers

The use of microphone arrays offers enhancements of speech signals recorded in meeting rooms and office spaces. A common solution for speech enhancement in realistic environments with ambient noise and multi-path propagation is the application of so-called beamforming techniques. Such beamforming algorithms enhance signals at the desired angle using constructive interference while attenuating signals coming from other directions by destructive interference. However, these techniques require as a priori the time difference of arrival information of the source. Therefore, the source localization and tracking algorithms are an integral part of such a system. The conventional localization algorithms deteriorate in realistic scenarios with multiple concurrent speakers. In contrast to conventional methods, the techniques presented in this paper make use of pitch information of speech signals in addition to the location information. This “position–pitch”-based algorithm pre-processes the speech signals by a multiband gammatone filterbank that is inspired from the auditory model of the human inner ear. The role of this gammatone filterbank is analyzed and discussed in details. For a robust localization of multiple concurrent speakers, a frequency-selective criterion is explored that is based on a study of the human neural system's use of correlations between adjacent sub-band frequencies. This frequency-selective criterion leads to improved localization performance. To further improve localization accuracy, an algorithm based on grouping of spectro-temporal regions formed by pitch cues is presented. All proposed speaker localization algorithms are tested using a multichannel database where multiple concurrent speakers are active. The real-world recordings were made with a 24-channel uniform circular microphone array using loudspeakers and human speakers under various acoustic environments including moving concurrent speaker scenarios. The proposed techniques produced a localization performance that was significantly better than the state-of-the-art baseline in the scenarios tested.     

A perceptually motivated stationary wavelet packet filterbank using improved spectral over-subtraction for enhancement of speech in various noise environments

In this paper, we propose a speech enhancement method where the front-end decomposition of the input speech is performed by temporally processing using a filterbank. The proposed method incorporates a perceptually motivated stationary wavelet packet filterbank (PM-SWPFB) and an improved spectral over-subtraction (I-SOS) algorithm for the enhancement of speech in various noise environments. The stationary wavelet packet transform (SWPT) is a shift invariant transform. The PM-SWPFB is obtained by selecting the stationary wavelet packet tree in such a manner that it matches closely the non-linear resolution of the critical band structure of the psychoacoustic model. After the decomposition of the input speech, the I-SOS algorithm is applied in each subband, separately for the estimation of speech. The I-SOS uses a continuous noise estimation approach and estimate noise power from each subband without the need of explicit speech silence detection. The subband noise power is estimated and updated by adaptively smoothing the noisy signal power. The smoothing parameter in each subband is controlled by a function of the estimated signal-to-noise ratio (SNR). The performance of the proposed speech enhancement method is tested on speech signals degraded by various real-world noises. Using objective speech quality measures (SNR, segmental SNR (SegSNR), perceptual evaluation of speech quality (PESQ) score), and spectrograms with informal listening tests, we show that the proposed speech enhancement method outperforms than the spectral subtractive-type algorithms and improves quality and intelligibility of the enhanced speech.     

Modulation Spectral Features for Robust Far-Field Speaker Identification

In this paper, auditory inspired modulation spectral features are used to improve automatic speaker identification (ASI) performance in the presence of room reverberation. The modulation spectral signal representation is obtained by first filtering the speech signal with a 23-channel gammatone filterbank. An eight-channel modulation filterbank is then applied to the temporal envelope of each gammatone filter output. Features are extracted from modulation frequency bands ranging from 3-15 H z and are shown to be robust to mismatch between training and testing conditions and to increasing reverberation levels. To demonstrate the gains obtained with the proposed features, experiments are performed with clean speech, artificially generated reverberant speech, and reverberant speech recorded in a meeting room. Simulation results show that a Gaussian mixture model based ASI system, trained on the proposed features, consistently outperforms a baseline system trained on mel-frequency cepstral coefficients. For multimicrophone ASI applications, three multichannel score combination and adaptive channel selection techniques are investigated and shown to further improve ASI performance.     

采用子带谱减法的语音增强

为了抑制语音信号中的环境噪声,提出了一种基于子带谱减法进行噪声抑制的语音增强方法。首先通过滤波器组将时域信号分成若干个频（子）带,然后在每个子带中,独立使用改进的谱减法技术进行语音增强。由于实际环境中的背景噪声绝大多数都不是随频率均匀分布的,因此这种在不同频带内进行噪声估计和频谱相减的方法更具有针对性,且更加准确。在实际语音处理实验中证明,所提方法在达到噪声抑制效果的同时较好地保留了语音的结构,使增强后的语音具有更高的听觉舒适度和可理解度。     

Robust feature extraction based on an asymmetric level-dependent auditory filterbank and a subband spectrum enhancement technique

In this paper we introduce a robust feature extractor, dubbed as robust compressive gammachirp filterbank cepstral coefficients (RCGCC), based on an asymmetric and level-dependent compressive gammachirp filterbank and a sigmoid shape weighting rule for the enhancement of speech spectra in the auditory domain. The goal of this work is to improve the robustness of speech recognition systems in additive noise and real-time reverberant environments. As a post processing scheme we employ a short-time feature normalization technique called short-time cepstral mean and scale normalization (STCMSN), which, by adjusting the scale and mean of cepstral features, reduces the difference of cepstra between the training and test environments. For performance evaluation, in the context of speech recognition, of the proposed feature extractor we use the standard noisy AURORA-2 connected digit corpus, the meeting recorder digits (MRDs) subset of the AURORA-5 corpus, and the AURORA-4 LVCSR corpus, which represent additive noise, reverberant acoustic conditions and additive noise as well as different microphone channel conditions, respectively. The ETSI advanced front-end (ETSI-AFE), the recently proposed power normalized cepstral coefficients (PNCC), conventional MFCC and PLP features are used for comparison purposes. Experimental speech recognition results demonstrate that the proposed method is robust against both additive and reverberant environments. The proposed method provides comparable results to that of the ETSI-AFE and PNCC on the AURORA-2 as well as AURORA-4 corpora and provides considerable improvements with respect to the other feature extractors on the AURORA-5 corpus.     

Fuzzy decision fusion of complementary experts based on evolutionary cepstral coefficients for phoneme recognition

Optimal representation of acoustic features is an ongoing challenge in automatic speech recognition research. As an initial step toward this purpose, optimization of filterbanks for the cepstral coefficient using evolutionary optimization methods is proposed in some approaches. However, the large number of optimization parameters required by a filterbank makes it difficult to guarantee that an individual optimized filterbank can provide the best representation for phoneme classification. Moreover, in many cases, a number of potential solutions are obtained. Each solution presents discrimination between specific groups of phonemes. In other words, each filterbank has its own particular advantage. Therefore, the aggregation of the discriminative information provided by filterbanks is demanding challenging task. In this study, the optimization of a number of complementary filterbanks is considered to provide a different representation of speech signals for phoneme classification using the hidden Markov model (HMM). Fuzzy information fusion is used to aggregate the decisions provided by HMMs. Fuzzy theory can effectively handle the uncertainties of classifiers trained with different representations of speech data. In this study, the output of the HMM classifiers of each expert is fused using a fuzzy decision fusion scheme. The decision fusion employed a global and local confidence measurement to formulate the reliability of each classifier based on both the global and local context when making overall decisions. Experiments were conducted based on clean and noisy phonetic samples. The proposed method outperformed conventional Mel frequency cepstral coefficients under both conditions in terms of overall phoneme classification accuracy. The fuzzy fusion scheme was shown to be capable of the aggregation of complementary information provided by each filterbank.     

Combining Auditory Preprocessing and Bayesian Estimation for Robust Formant Tracking

We present a framework for estimating formant trajectories. Its focus is to achieve high robustness in noisy environments. Our approach combines a preprocessing based on functional principles of the human auditory system and a probabilistic tracking scheme. For enhancing the formant structure in spectrograms we use a Gammatone filterbank, a spectral preemphasis, as well as a spectral filtering using Difference-of-Gaussians (DoG) operators. Finally, a contrast enhancement mimicking a competition between filter responses is applied. The probabilistic tracking scheme adopts the mixture modeling technique for estimating the joint distribution of formants. In conjunction with an algorithm for adaptive frequency range segmentation as well as Bayesian smoothing an efficient framework for estimating formant trajectories is derived. Comprehensive evaluations of our method on the VTR–Formant database emphasize its high precision and robustness. We obtained superior performance compared to existing approaches for clean as well as echoic noisy speech. Finally, an implementation of the framework within the scope of an online system using instantaneous feature-based resynthesis demonstrates its applicability to real-world scenarios.      

Filterbank optimization for robust ASR using GA and PSO

Automatic speech recognition (ASR) systems follow a well established approach of pattern recognition, that is signal processing based feature extraction at front-end and likelihood evaluation of feature vectors at back-end. Mel-frequency cepstral coefficients (MFCCs) are the features widely used in state-of-the-art ASR systems, which are derived by logarithmic spectral energies of the speech signal using Mel-scale filterbank. In filterbank analysis of MFCC there is no consensus for the spacing and number of filters used in various noise conditions and applications. In this paper, we propose a novel approach to use particle swarm optimization (PSO) and genetic algorithm (GA) to optimize the parameters of MFCC filterbank such as the central and side frequencies. The experimental results show that the new front-end outperforms the conventional MFCC technique. All the investigations are conducted using two separate classifiers, HMM and MLP, for Hindi vowels recognition in typical field condition as well as in noisy environment.