基于感知加权滤波的卡尔曼滤波语音增强

结合人耳听觉掩蔽效应,提出一种基于听觉感知加权的卡尔曼滤波语音增强方法。由于人耳对语音的感知主要是通过语音信号频谱分量幅度获得的,引入听觉感知加权滤波器在频域上使共振峰区域残留噪声更多,而共振峰之间及语音幅度谱较低的区域残留噪声减少,这样符合人耳的听觉特性,从而使得主观感觉到的噪声最小。采用语音质量感知评估对语音增强的效果进行评测,与传统的卡尔曼滤波语音增强算法相比,实验结果显示该算法提高了增强语音的质量。     

掩蔽效应在有源噪声控制中的应用

提出了一种通过频域内的加权手段,将心理声学模型的掩蔽效应特性应用到有源噪声控制系统中的有源噪声控制方法。详细给出了将掩蔽效应特性具体应用到有源噪声控制中的实现方法。仿真结果证明了这种新的有源噪声控制方法降低了噪声的分贝量,改善了频谱特性,提高了残余噪声的主观性能。     

相控阵噪声系数特性及其典型应用分析

针对相控阵接收系统双层合成网络的一般架构,逐层分解并简化分析得出相控阵接收系统等效接收增益和噪声系数两个重要特性的通用表达式。同时,分析了低噪放、损耗、幅度加权等环节对噪声系数特性的重要影响及其工程设计要点,并结合雷达系统工程设计给出相控阵噪声系数特性五种典型应用的举例和计算评估分析,验证了之前幅度加权等因素对系统噪声系数影响的分析结论。     

Two‐Microphone Generalized Sidelobe Canceller with Post‐Filter Based Speech Enhancement in Composite Noise

This paper describes an algorithm to suppress composite noise in a two‐microphone speech enhancement system for robust hands‐free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal‐dominant time‐frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech‐dominant TFBs are identified among the previously detected nonstationary signal‐dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin‐wise output signal‐to‐noise ratio is obtained with these power estimates and a Wiener post‐filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post‐filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.     

双重插值法噪声频谱计权计算及其误差分析

为实现对采集到的噪声信号进行声压级计算,提出了一种新的基于样条插值法的频谱计权算法。与传统频谱计权方法比较,该方法计算精度高,且便于计算机实现。误差分析结果显示,用该方法进行A计权噪声频谱计权计算的绝对误差总和小于0．0843dB,相对误差总和小于1．25％,最大相对误差小于0．436％。     

Efficient ICI Self‐Cancellation Scheme for OFDM Systems

In this paper, we present a new inter‐carrier interference (ICI) self‐cancellation scheme — namely, ISC scheme — for orthogonal frequency‐division multiplexing systems to reduce the ICI generated from phase noise (PHN) and residual frequency offset (RFO). The proposed scheme comprises a new ICI cancellation mapping (ICM) scheme at the transmitter and an appropriate method of combining the received signals at the receiver. In the proposed scheme, the transmitted signal is transformed into a real signal through the new ICM using the real property of the transmitted signal; the fast‐varying PHN and RFO are estimated and compensated. Therefore, the ICI caused by fast‐varying PHN and RFO is significantly suppressed. We also derive the carrier‐to‐interference power ratio (CIR) of the proposed scheme by using the symmetric conjugate property of the ICI weighting function and then compare it with those of conventional schemes. Through simulation results, we show that the proposed ISC scheme has a higher CIR and better bit error rate performance than the conventional schemes.     

基于SeeSV噪声定位技术的电机噪声测试

针对驱动电机某一运转工况的噪声声压级与电机噪声频率的测试问题,提出了一种运用声学照相机SeeSV与传声器相结合,并同时检测电机振动信号的测试方法。通过选取SeeSV的不同频段对噪声源进行定位识别,分离出驱动电机的噪声频段,运用传声器测定电机的噪声强度,并用振动信号对声压信号进行验证。测试结果表明,用SeeSV和传声器相结合的方法,可在复杂噪声中识别出驱动电机的噪声频率为2 66625 Hz,声压级为607 dB,对工程噪声的评价具有实践意义。     

用System View滤除伴音中的窄带噪声

有效滤除伴音中的噪声对提高节目播出质量是有帮助的。System View是一个功能强大的数字信号处理和通信系统设计的仿真软件，它提供了信号从时域到频域的转换和滤波器的设计工具，方便捕捉声音信号中的噪声频谱并通过设计合理的滤波器滤除噪声。本例说明使用System View软件滤除伴音信号中的窄带噪声的方法。     

基于延迟LMS算法的管道噪声有源控制实验研究

在自适应有源控制中，次级通路建模对控制器算法实现有着重要的影响。以往次级通路建模通常会使用带有数十个甚至上百个系数的FIR滤波器，但过多的参数运算会对系统稳定性、控制实时性及硬件复杂度等方面带来麻烦。研究了一种在工程中实现简便的次级通路建模方法——通路时延估计法，并在消声室内进行了管道噪声有源控制实验。实验结果表明，次级通路时延影响着管道噪声有源控制的降噪频率以及降噪带宽。     

自适应主动噪声控制系统的仿真研究

对主动噪声控制系统的前馈型、反馈型和混合型3种控制方式进行了比较,利用Simulink设计了实现FxLMS算法的模块,并进行仿真。结果表明,前馈型控制方式可降低与参考信号相关的噪声;反馈型控制方式可降低可预测的噪声;而混合型控制方式可同时降低这2种噪声。     

Audio Transcoding for Audio Streams from a T‐DTV Broadcasting Station to a T‐DMB Receiver

We propose an efficient audio transcoding algorithm that can convert audio streams from terrestrial digital television broadcasting service stations to those for terrestrial digital multimedia broadcasting hand‐held receivers. The proposed algorithm avoids the complicated psychoacoustic analysis by calculating the scalefactors of the bit‐sliced arithmetic coding encoder directly from the signal‐to‐noise ratio parameters of the AC‐3 decoder. The bit‐allocation process is also simplified by cascading the nested distortion control loop. Through subjective evaluation, it is shown that the proposed algorithm provides comparable audio quality to tandem coding but it requires much smaller complexity.     

基于遗传算法的抽指加权叉指换能器进化设计

提出了极性抽指加权叉指换能器设计的新方法，克服了传统设计方法的繁杂性。将极性抽指加权叉指换能器作为染色体，通过独特的（-1，1）的二值编码，以目标频率响应曲线和待进化的叉指换能器频率响应曲线在考虑的频率范围内的1601个采样点的误差值为进化目标，对种群中的染色体进行选择、交叉和变异等遗传操作，自动进化出符合目标要求的极性抽指加权叉指换能器极性加权状况。进化实验结果表明，应用本文提出的进化方法设计出的极性抽指加权叉指换能器的频响曲线与目标频响曲线基本重合，达到设计要求，进化设计方法效率高，实用性强。     

一种基于感知滤波的语音去噪算法

文中针对加性白噪声的环境下,通过在传统的Wiener滤波算法中引入人耳听觉感知特性及谱减算法,提出了一种新的基于感知滤波的语音去噪算法.该算法的关键是采用LPC分析得出的感知加权函数修正维纳滤波方程,使噪声谱分布随语音而变.不仅保持Wiener滤波算法的优点,而且降低噪声对纯净语音的影响.实验表明,该方法能更有效地抑制背景噪声,提高语音质量,且具有较低的计算复杂度.     

语言录（播）音室和审听室的声学特性──中国国际广播中心的部分声学设计

文章论述关于语录室和审听室的噪声振动隔声的措施及应达到和可能达到的效果。并以波动声学的观点讨论在＂小房间＂中的音质问题。并就低频吸声处理及声染色等与具体声学设计的关系提出看法。文中还就审听室声学设计的一些概念进行讨论。     

基于模糊归一化神经网络的开环子带主动消噪

将无延时开环子带自适应滤波结构应用到传统的宽带主动消噪系统中,并提出开环子带结构中的X滤波预补偿方法,弥补了滤波器延迟对系统的影响,提高了系统的收敛速度,降低了计算复杂度。此外,利用神经网络来精细模拟实际噪声的高度非线性性,提出模糊归一化收敛步长调整方法以控制开环方案带来的额外误差能量,并给出了数学证明。对实际噪声的仿真结果显示,给出的主动消噪系统和算法具有更快的收敛速度和更低的稳态误差,达到了更好的降噪效果。     

Two‐Microphone Binary Mask Speech Enhancement in Diffuse and Directional Noise Fields

Two‐microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time–frequency (T‐F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T‐F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T‐F units with hit rates above 85%. It outperforms previous solutions in terms of signal‐to‐noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.     

Fully Differential 5‐GHz LC‐Tank VCOs with Improved Phase Noise and Wide Tuning Range

In this paper, we propose two LC voltage‐controlled oscillators (VCOs) that improve both phase noise and tuning range. With both 1/f induced low‐frequency noise and low‐frequency thermal noise around DC or around harmonics suppressed significantly by the employment of a current‐current negative feedback (CCNF) loop, the phase noise in the CCNF LC VCO has been improved by about 10 dB at 6 MHz offset compared to the conventional LC VCO. The phase noise of the CCNF VCO was measured as ?112 dBc/Hz at 6 MHz offset from 5.5 GHz carrier frequency. Also, we present a bandwidth‐enhanced LC VCO whose tuning range has been increased about 250 % by connecting the varactor to the bases of the cross‐coupled pair. The phase noise of the bandwidth‐enhanced LC‐tank VCO has been improved by about 6 dB at 6 MHz offset compared to the conventional LC VCO. The phase noise reduction has been achieved because the DC‐decoupling capacitor Cc prevents the output common‐mode level from modulating the varactor bias point, and the signal power increases in the LC‐tank resonator. The bandwidth‐enhanced LC VCO represents a 12 % bandwidth and phase noise of ?108 dBc/Hz at 6 MHz offset.     

Adaptive noise cancellation in a multimicrophone system for distortion product otoacoustic emission acquisition

This study focuses on adaptive noise cancellation (ANC) techniques for the acquisition of distortion product otoacoustic emissions (DPOAEs). Otoacoustic emissions (OAEs) are very low level sounds produced by the outer hair cells of normal cochleas, spontaneously or in response to sound stimulation as a byproduct of a frequency and threshold sensitivity increasing mechanism. Current OAE recording systems rely on test probe noise attenuation and synchronous ensemble averaging for increasing signal-to-noise ratios (SNRs). The efficiency of an ANC algorithm for noise suppression was investigated using three microphones: one placed in the test ear, one in the nontest ear for internal noise reference; one near the subject's head for external noise reference. The system proposed was tested with simulations, off-line averaging and real-time implementation of the ANC algorithm. Simulation results showed that the technique had a potential noise reduction capability of 24 dB for complex multifrequency noise signals. Off-line results were positive, with a mean SNR improvement of 4.9 dB. Real-time results indicated that the use of an ANC algorithm in combination with standard averaging methods can reduce noise levels by as much as 10 dB beyond that obtained with standard noise reduction methods and probe attenuation alone.     

Controller design for active noise cancellation headphones usingexperimental raw data

This work presents a practical method of designing controllers for active noise cancellation (ANC) headphones. Without attempting system identification and perturbation modeling, the headphone system is directly described by a set of frequency-response data. In frequency domain, the controller synthesis problem is formulated as a constrained optimization problem, where the H₂ performance objective is minimized with various frequency-dependent constraints. The fixed-order robust controller is thus designed to achieve maximum noise attenuation with acceptable stability margins. Further, the method is able to accurately constrain noise amplification outside the control bandwidth due to waterbed effect of the nonminimum phase plants. This feature is very important for the ANC design to maintain an overall quality of noise reduction     

A Novel Voice Coder At 4800BPS(HSEV)

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