An OFDM all digital in-band-on-channel (IBOC) AM and FM radiosolution using the PAC encoder

The advances in digital communications and compression algorithms have made more efficient and more robust transmission schemes possible. Radio broadcast systems have not fully utilized these advances to their benefit. All digital robust radio broadcast systems for the AM and the FM band are proposed. The proposed systems are based on orthogonal frequency division multiplexing (OFDM) technology in conjunction with PAC for both the AM and the FM bands. The Perceptual Audio Coder (PAC) developed by Bell Laboratories compresses audio signals very efficiently with CD-like quality at 96 kbps and stereo FM-like quality at 48 kbps. These are rates achievable with 200 kHz and 30 kHz bandwidths available per FM and AM station respectively. No new spectrum is required since the digital signals are transmitted within the current allocated FCC masks. In an FM channel, a wide-band data subchannel can be provided in addition to the 96 kbps error protected PAC audio information     

Distributed audio sensing with homeostasis-inspired autonomous communication

Emerging applications of wireless sensor networks (WSN) requiring wide-band event signal communication such as multimedia surveillance sensor networks impose additional challenges including high communication bandwidth requirement and energy cost. Besides their partially or fully dependency on feedback messages from sink node, the existing protocols designed for WSN do not address the communication of wide-band event signals. Furthermore, the feedback messages may not reach in time to provide reliable communication of event information and save scarce network resources. Therefore, an autonomous communication protocol is imperative in order to provide wide-band event signal communication without any feedback from the sink. In nature, biological systems have self-organization capability, i.e., homeostasis, as they autonomously maintain a relatively stable equilibrium state for operation of vital functions. Hence, this natural phenomenon clearly gives promising inspirations in order to develop autonomous and efficient communication models and protocols for WSN domain. In this paper, the homeostasis-inspired autonomous communication (HAC) protocol is introduced for wireless audio sensor networks (WASN). Using the spectral properties of the wide-band event signal, i.e., audio signal, HAC enables WASN to maintain a relatively stable state in which sensor nodes reliably and energy-efficiently communicate the event signal to the sink node. Furthermore, with its self-organization capability, HAC does not rely on any feedback message from the sink node. Performance evaluations reveal that HAC successfully communicates wide-band event signal with minimum energy expenditure.     

Low bit rate transparent audio compression using adapted wavelets

Describes a novel wavelet based audio synthesis and coding method. The method uses optimal adaptive wavelet selection and wavelet coefficients quantization procedures together with a dynamic dictionary approach. The adaptive wavelet transform selection and transform coefficient bit allocation procedures are designed to take advantage of the masking effect in human hearing. They minimize the number of bits required to represent each frame of audio material at a fixed distortion level. The dynamic dictionary greatly reduces statistical redundancies in the audio source. Experiments indicate that the proposed adaptive wavelet selection procedure by itself can achieve almost transparent coding of monophonic compact disk (CD) quality signals (sampled at 44.1 kHz) at bit rates of 64-70 kilobits per second (kb/s). The combined adaptive wavelet selection and dynamic dictionary coding procedures achieve almost transparent coding of monophonic CD quality signals at bit rates of 48-66 kb/s     

自适应天线阵列在宽带CDMA上行链路接收机中的应用

本文提出了一种将自适应天线阵列与横向均衡器相结合。采用局部基于码片速率的信道估计和可变步长自适应算法的宽带CDMA上行链路空-时接收机结构。通过理论分析并结合CDMA2000标准上行链路的计算机仿真,证明这种空-时接收机在瑞利衰落信道下,具有较好的收敛性能和接收性能。     

空时自适应抗干扰算法分析与仿真

针对卫星信号易受干扰的问题，阐述了空时二维滤波算法在自适应波束形成系统中的应用，并对几种常用的自适应处理算法进行了简单介绍，最后，给出一组宽带信号的数据，运用以上几种算法在仿真系统中分析比较它们抗干扰性能。     

A RAKE receiver design for WCDMA FDD uplink with an RLS-based adaptive beamforming scheme

In this paper, we present a RAKE receiver design with adaptive antenna arrays for the wide-band code-division multiple-access (WCDMA) frequency-division duplexing (FDD) uplink. The RLS-based adaptive beamforming scheme is proposed and can be built with the existing one-dimensional RAKE receiver. We adaptively calculate the beamforming weight vector for each multipath of the desired user, and use maximum ratio combining (MRC) to combine each multipath signal in the demodulation process. Two matched filters based on the spreading waveforms are designed in our scheme for WCDMA FDD uplink application. The proposed scheme has the ability of suppressing strong multiuser access interference and the other types of interferers through spatial ing. The tracking capability of the proposed algorithm is demonstrated in the simulation results.     

基于MPEG-AAC编码器的压缩域音频增强方法

本文基于MPEG-AAC音频编解码器,提出了一种压缩域的音频增强方法.首先,对含噪音频信号的比特流进行解码,得到含噪音频信号的MDCT系数;然后,利用修正的加权递归平均（Modified Weighted Recursive Averaging,MWRA）方法估计噪声功率;再者,利用基于听觉掩蔽原理的自适应β-阶双曲余弦（COSH）统计模型,对含噪音频的MDCT系数进行增强处理;最后,将增强后的MDCT系数重新量化编码,得到用于解码的增强比特流实验结果表明,本文提出的方法能有效去除AAC解码音频信号中的多种背景噪声,其性能明显优于参考方法.     

Effect of jammer power on the performance of adaptive arrays

The effect of jammer power on the performance of adaptive arrays is studied. It is shown that the output signal-to-interference-plus-noise ratio (SINR) of an adaptive array is a function of jammer power. In the presence of a wide-band jammer, the output SINR of the array decreases with an increase in jammer power and eventually goes to zero. Unlike continuous wave (CW) jammers, a wide-band jammer does not go through power inversion. Instead, as the jammer power is increased, the interference-to-noise ratio (INR) at the array output shows oscillations. For large jammer power, the output INR increases with an increase in jammer power.     

基于能量特性分块的DCT域自适应音频水印算法

提出了一种基于能量特性分块的DCT域自适应音频水印算法。首先对音频数据做DCT变换,然后基于子块占总能量的比例将相应的频域系数分成几个部分,用每一部分的能量特性系数构造水印的嵌入强度因子,从而实现自适应的嵌入水印。实验结果表明：该算法对数字音频信号的噪声攻击、低通滤波、MPEG压缩、D/A和A/D转换等操作具有很强的鲁棒性。     

Comment on “Constrained Hung-Turner adaptive beamformingalgorithm with additional robustness to wide-band and movingjammers” [and reply]

One of the major claims in the paper of Gershman et al. (see ibid., vol.44, p.361-6, 1996) is that Hung-Turner (1983) projections do not adequately suppress wide-band interference. This statement is only true if too few snapshots are used to estimate the interference subspace. For Hung-Turner projections the use of more snapshots than the rank of the interference is called “averaging over the dimension”. The authors implicitly and wrongly claim that even with “averaging over the dimension”, Hung-Turner projections perform poorly in the presence of wide-band interference. In this comment, we show theoretically and by simulation that Hung-Turner projections adequately suppress wide-band jammers if the number of snapshots is equal to or larger than the rank of the interference. Here, the same sensor array processing model and assumptions are used. Gershman et al. reply that they proposed a new modification of the Hung-Turner (HT) algorithm improving the robustness of this algorithm against wide-band and moving jammers. They also add that their approach can be applied to other adaptive array algorithms as well, and the critical comment contains a misinterpretation of their results and conclusions     

多业务无线蜂窝移动通信系统的一种呼叫允许控制策略

第三代移动通信系统要求支持宽带多媒体业务,如话音、视频、数据等多种业务,不同业务有不同的QoS要求。本文提出的多业务无线蜂窝移动通信系统中一种基于QoS的呼收允许控制策略,对不同业务的切换呼叫给予不同的优先权。本文分析了两种呼叫允许控制（CAC）算法,一种是各种业务的切控呼叫无缓冲器,不进入排队系统;另一种是各种业务的切换呼叫设置有缓冲器,进入排除系统,并且话音、视频业务的切切呼叫比数据业务的切换呼叫有更高的优先权,系统的空闲信道应首先分配给话音、视频业务的切换呼叫,再分配给数据业务的切换呼叫。在分析两种CAC算法的呼叫阻塞概率、切换失败概率以及系统吞吐量的基础上,给出了计算机仿真结果。     

阵列天线波束成形技术在宽带CDMA下行链路中的应用

本文利用自适应天线阵列上行链路信号到达角估计信息,并结合导频辅助下行链路信道矩阵估计值,在前人提出的具有反馈特性的自适应发送阵列原理基础上,提出一种简单可行的宽带CDMA下行链路天线阵列波束成形权重计算方法,并进一步分析了到达角估计误差和信道信息反馈时延对权重计算的影响,最后给出系统误码率性能的计算机仿真结果。     

A Bandpass Filter With Inherent Gain Adaptation for Hearing Applications

In this paper, we propose a novel bandpass filter design that incorporates automatic gain control (AGC). The gain control in the filter reduces the performance requirements of a wide-band AGC, and allows for low-power multichannel compression. The filter achieves up to 15 dB of compression on a 55-dB input dynamic range and is tunable over the audio frequency range, with microwatt power consumption and <5% harmonic distortion.     

自适应捷变频雷达干扰技术分析

介绍了自适应捷变频雷达抗干扰技术特征，讨论了自适应捷变频雷达的几种干扰方法：欺骗自适应干扰、回答式干扰、扫频式干扰和宽带阻塞干扰。     

New Low Bit-Rate Audio Coding Approach Based on Adaptive Wavelet Analysis

This paper describes a new audio coding scheme based on adaptive wavelet analysis that provides transparent audio coding for CD-audio signals at low bit rates (≈1.4 bits/sample per channel). A new perceptual cost function is defined to obtain the best wavelet-packet base for each audio frame. The sharp variations in quantization noise that appear at the border of the frames are minimized by a novel approach that avoids overlapping. The proposed coder guarantees high perceptual quality using filters that generate wavelets of any compact support, because a bit-allocation algorithm that takes into account the equivalent filter frequency responses of the synthesis filter bank branches is used.     

一种自适应语音增强均衡器的研究

在参考滤波器组的基础上，提出一个用于语音和音频信号进行时不变或自适应谱修正的数字滤波器结构，主要用于音频信号均衡和降噪。在频域计算滤波器系数的同时，信号在时域滤波。与通常频域处理相比，在信号延迟、原型滤波器设计、复杂性等方面，该结构有良好的特性。该算法既适用于均匀频率分辨率，也适用于非均匀频率分辨率。     

参数音频编码算法的改进

参数音频编码是一种基于产生音频信号的参数模型的有效编码方法。最近已用于很低码率的音频编码中,如MPEG-4Version2音频标准HILN。这里提出HILN参数音频编码的改进算法。改进算法采用了自适应的基于倒谱的基频检测算法,利用倒谱和频谱特性,保证了大范围内基频提取的效率和准确性。在正弦波合成中采用相位匹配合成算法,保证了帧间参数的连续性。改进算法还对残差信号进行预处理,去除不必要的干扰分量,最大限度重建原音频信号,提高了合成音频信号的质量。     

A CMOS digitally controlled audio attenuator for Hi-Fi systems

Describes an audio attenuator designed for use in digitally controlled Hi-Fi systems. The circuit is fabricated with a thin film on CMOS process and has a dynamic range of 0 to -88.5 dB in 1.5 dB steps. Total harmonic distortion for the complete circuit is typically -87 dB and wide-band signal/noise ratio is typically -86 dB.     

基于频域模型的宽带信号子空间谱估计方法

宽带阵列信号是频率的函数,因此其阵列流型及协方差矩阵都随频率变化.本文基于宽带信号的频域模型,通过分析宽带阵列信号协方差矩阵的特征分解结构,证明了宽带阵列信号噪声子空间不随频率变化的特性,并根据这一特性,提出了基于频域模型的宽带子空间谱估计(SSEFD)方法.应用K.Buckley的BASS-ALE方法解决了该方法用于均匀线阵时存在的频率-方位模糊问题.计算机仿真结果验证了SSEFD方法的有效性,与H.Wang的CSS方法的统计性比较表明,新方法具有更高的估计精度.