Robust Audio Fingerprinting Method Using Prominent Peak Pair Based on Modulated Complex Lapped Transform

The robustness of an audio fingerprinting system in an actual noisy environment is a major challenge for audio‐based content identification. This paper proposes a high‐performance audio fingerprint extraction method for use in portable consumer devices. In the proposed method, a salient audio peak‐pair fingerprint, based on a modulated complex lapped transform, improves the accuracy of the audio fingerprinting system in actual noisy environments with low computational complexity. Experimental results confirm that the proposed method is quite robust in different noise conditions and achieves promising preliminary accuracy results.     

Online Blind Channel Normalization Using BPF‐Based Modulation Frequency Filtering

We propose a new bandpass filter (BPF)‐based online channel normalization method to dynamically suppress channel distortion when the speech and channel noise components are unknown. In this method, an adaptive modulation frequency filter is used to perform channel normalization, whereas conventional modulation filtering methods apply the same filter form to each utterance. In this paper, we only normalize the two mel frequency cepstral coefficients (C0 and C1) with large dynamic ranges; the computational complexity is thus decreased, and channel normalization accuracy is improved. Additionally, to update the filter weights dynamically, we normalize the learning rates using the dimensional power of each frame. Our speech recognition experiments using the proposed BPF‐based blind channel normalization method show that this approach effectively removes channel distortion and results in only a minor decline in accuracy when online channel normalization processing is used instead of batch processing.     

Pre‐filtering technique for MAI cancellation in MC‐CDMA systems

Multicarrier code division multiple access (MC‐CDMA), is a promising multiplexing technique for future communication systems. In this study, we employ the well‐known Walsh‐Hadamard spreading codes for synchronous downlink transmission of MC‐CDMA systems. The spreading codes allow that the frequency diversity to be efficiently exploited. However, multipath propagation may cause orthogonality among users is distorted, and this distortion produces multiple access interference (MAI). To eliminate this effect, we propose a pre‐filtering‐based MC‐CDMA system which uses a pre‐filtering technique at the transmitter and an equal gain combining (EGC) scheme at the receivers, respectively. Our proposed pre‐filtering technique transforms the transmitted signals so that the MAI can be eliminated, and the EGC scheme weights the signals received from all subcarriers so that channel distortions can be compensated. Furthermore, the proposed technique can calculate the transmitted power over all subcarriers to satisfy the required quality of service of each user and archive MAI‐free. In this paper, performance in terms of bit error rate is analyzed; in comparison with the EGC, orthogonal restoring combining, and maximal ratio combining schemes at receiver, respectively. Copyright © 2011 John Wiley & Sons, Ltd.     

一种小波域新颖鲁棒语音扩谱水印

文章提出一种基于小波变换的新颖的鲁棒语音扩谱水印算法。算法对原始语音进行离散小波变换．利用扩谱水印技术将水印隐藏到小波域。根据检测与估计理论，运用新的检测统计量进行相关检测。同时，通过引入抗异步攻击的机制和倒谱滤波，增强了算法的鲁棒性。实验结果表明，该算法对噪声、中值滤波、低通滤波、异步攻击等有较强的鲁棒性。     

Two‐pilot channel estimation scheme for amplify‐and‐forward relay networks

In amplify‐and‐forward relay networks, the equivalent channel to the destination node is not independent of equivalent noise and the equivalent noise does not follow a Gaussian distribution. Therefore, it is difficult to directly estimate the equivalent channel based on traditional optimal rules. In this paper, we propose a two‐pilot estimation (TPE) scheme that decomposes a non‐Gaussian noise channel estimation problem into two channel estimation problems in Gaussian noise. In TPE scheme, the relay‐destination channel is first estimated by one pilot and the other pilot is used to estimate the equivalent channel with the aid of the estimated relay‐destination channel. Simulation results show that the TPE scheme can achieve less estimation error and larger system throughput than other existing channel estimators in slow fading case. Copyright © 2012 John Wiley & Sons, Ltd.     

A parameter estimation method for time‐frequency‐overlapped frequency hopping signals based on sparse linear regression and quadratic envelope optimization

The frequency hopping (FH) signals have well‐documented merits for commercial and military fields due to near‐far resistance and robustness to jamming. Therefore, the parameter estimation of FH signals is an important task for subsequent information acquisition and autonomous electronic countermeasure or attack. However, under the complex electromagnetic environment, there always exist overlaps in the time‐frequency domain among multiple signals, which bring poor signal sparsity and make the estimation more challenging. In this paper, a novel parameter estimation approach is developed for the time‐frequency‐overlapped FH signals under single‐channel reception. The exact solution is mainly composed of the sparse linear regression‐based matrix optimization (SLR‐MO) and quadratic envelope optimization (QEO). SLR‐MO highlights the removal of noise and distortion features for improving the overall sparsity and time‐frequency resolution. QEO further eliminates parts of the interfering signal features and outliers and then extracts and optimizes the average time‐frequency ridge to complete the parameter estimation (hopping instants, period, and carriers). Simulation results demonstrate that the developed estimator outperforms the traditional methods in the scope of application, estimation accuracy, and the robustness under low signal‐to‐noise ratio (SNR).     

基于提升小波的多重数字音频水印

该文提出了一种在提升小波域同时嵌入鲁棒水印和易损水印的音频水印算法。原始音频信号经过提升小波变换后,将低频小波系数进行均值量化嵌入鲁棒水印,具有较好的鲁棒性和不可感知性;对高频小波系数直接进行单个系数量化嵌入易损水印,当音频内容发生篡改时,这些水印信息会发生相应的改变,从而可以鉴定原始音频的完整性。水印的提取不需要原始音频信号。实验结果表明,鲁棒水印对MP3压缩、低通滤波、加噪、重量化、重采样等信号处理攻击具有很强的鲁棒性;而易损水印对上述攻击则具有很强的敏感性。     

PWM调制波的多层感知神经网络音频识别

研究了在PWM调制情形下音频信号的保真度分析及音频信号的频率、相位分布建模及其识别.基于Matlab/Simulink建模环境下建立了音频识别系统,利用滤波器对调制波进行归一化操作来完成对信号的预处理.基于音频信号模型构造了3类音频信号.经过识别系统的调制、滤波调理后输入至训练好的多层感知神经网络,分析识别结果显示,音频频率越高时,载波频率较低会降低系统识别率;提高载波频率可提高系统的识别率;并与相应的经典音频识别方案进行了比较.     

Multi‐channel Audio Service in a Terrestrial‐DMB System Using VSLI‐Based Spatial Audio Coding

Spatial audio coding (SAC) is an extremely high compact representation of encoded multi‐channel audio material. This paper suggests a multi‐channel audio service in the terrestrial digital multimedia broadcasting (T‐DMB) system using a novel SAC tool, which is called a virtual source location information (VSLI)‐based SAC tool. Intensive experiments are presented to evaluate the validity of the proposed VSLI‐based SAC tool, and prototypical systems are also presented to demonstrate the reliability of the proposed multi‐channel T‐DMB system in real applications.     

On‐line wavelet filtering of narrowband noise in signal detection of spread spectrum system for location tracking

Spread spectrum signal transmitted by wireless channel for location tracking can be severely corrupted by noise due to external disturbances. Narrowband noise is the most effective interference that can make measurement signal undetected. However, the current methods for narrowband interference (NBI) suppression are either very time‐consuming or add distortion to the signal received. In this paper, an adaptive Gaussian wavelet filter with optimal time–frequency localization and variable notch depth is proposed to suppress a large number of NBIs with additive white Gaussian noise and pulsed noise that interfere with the spread spectrum communication system. The filtering of both continuous and time‐varying NBIs with fast resampling is performed in conjunction with the fast Fourier transform‐based correlation for peak detection, and is computationally efficient for real‐time operation of signal detection. The performance of the adaptive filter has been evaluated by experiments employing a reliable noise detector. Experimental results demonstrate that the proposed wavelet filter isolates the signals from the NBI in accordance with the corrupted frequency contents while preserving the desired spread spectrum signal, and improves signal to noise ratio for peak detection leading to higher accuracy of timing measurement for real‐time wireless location. Copyright © 2011 John Wiley & Sons, Ltd.     

一种新的扩频音频水印算法

音频水印提供了一种数字保密的方法，用来保护作者和版权所有者的权利。在本文中。基于扩频技术提出了一种新的音频水印算法．即应用音频的瞬时平均频率(IMF)来嵌入数字水印，其目标是最大限度的满足水印的不可觉察性和健壮性。本文算法对原始的音频信号进行短时傅立叶变换，从而估计出信号的加权瞬时平均频率(IMF)。基于心理声学模型的掩蔽特性。可以得出水印被要求的相应的声压级。根据这些结果调制产生一个依赖于信号的不可觉察的水印。本算法允许在5秒钟的音频信号中嵌入25位信息。实验结果表明本文算法对于常见的信号处理攻击包括滤波、MP3压缩和添加噪声具有很好的健壮性。     

Low‐Power‐Adaptive MC‐CDMA Receiver Architecture

This paper proposes a novel concept of adjusting the hardware size in a multi‐carrier code division multiple access (MC‐CDMA) receiver in real time as per the channel parameters such as delay spread, signal‐to‐noise ratio, transmission rate, and Doppler frequency. The fast Fourier transform (FFT) or inverse FFT (IFFT) size in orthogonal frequency division multiplexing (OFDM)/MC‐CDMA transceivers varies from 1024 points to 16 points. Two low‐power reconfigurable radix‐4 256‐point FFT processor architectures are proposed that can also be dynamically configured as 64‐point and 16‐point as per the channel parameters to prove the concept. By tailoring the clock of the higher FFT stages for longer FFTs and switching to shorter FFTs from longer FFTs, significant power saving is achieved. In addition, two 256 sub‐carrier MC‐CDMA receiver architectures are proposed which can also be configured for 64 sub‐carriers in real time to prove the feasibility of the concept over the whole receiver.     

Channel estimation algorithms for cooperative spectrum sensing in amplify‐and‐forward cooperative system

In amplify‐and‐forward (AF)‐based cooperative spectrum sensing system, the bit‐error‐rate (BER) performance and detection probability will decrease because of the existence of channel estimation error. In this paper, the influence of channel estimation error on system performance is firstly deduced, and then, linear minimum mean‐square error (LMMSE) channel estimation algorithm with filtering delay time‐domain windowing (LMMSE‐filtering‐DTW) technique and modified singular value decomposition‐based LMMSE algorithm are proposed to improve the channel estimation performance for code division multiple access system and orthogonal frequency division multiplexing system in AF cooperative scenario, respectively. Simulation results verify the effectiveness of the two proposed channel estimation algorithms in cooperative spectrum sensing, and when E_b/ N₀ is bigger than 20 dB, given the required false alarm probability smaller than 15%, the difference of detection probability between the channel obtained using the proposed channel estimation algorithms and the ideal channel is less than 2.5%, respectively. Copyright © 2012 John Wiley & Sons, Ltd.     

Time‐domain equalizer for multicarrier systems in impulsive noise

In multicarrier communication systems, a time‐domain equalizer (TEQ) can be applied to shorten the channel impulse response and to eliminate inter‐symbol interference (ISI). However, the presence of impulsive noise in the channel may paralyze the operation of TEQs and subsequently lead to poor error performance. In this paper, a multicarrier receiver that incorporates a constant false alarm rate algorithm and an iterative estimation technique (CFAR‐IET) in conjunction with a TEQ is proposed to increase the robustness of the receiver against impulsive noise. Furthermore, an improved version of the CFAR‐IET‐TEQ scheme, which uses the buffering, sorting, removing and amplitude averaging (BSRA) processes, is presented. Performance comparisons of the proposed schemes with the existing Gaussian‐optimized schemes are made. Simulation results show that the BSRA‐IET‐TEQ scheme is an effective approach to reduce symbol error rate (SER) in impulsive channels while performing satisfactorily in Gaussian channels. Copyright © 2011 John Wiley & Sons, Ltd.     

An Efficient Time‐Frequency Representation for Parametric‐Based Audio Object Coding

Object‐based audio coding can provide new music applications with interactivity. To efficiently compress a lot of target audio objects, a subband‐based parametric coding scheme has been adopted for MPEG spatial audio object coding. In this letter, the time‐frequency (T/F) subband analysis structure is investigated. A reconfigured T/F structure is also proposed to enhance the generating performance of sound scenes such as ‘karaoke’ and ‘solo’ play in interactive music scenarios. From the experimental results, it was confirmed that the proposed scheme remarkably improves the SNR and sound quality.     

Perceptual audio data concealment and watermarking scheme using direct frequency domain substitution

A method is described for perceptually transparent data concealment and watermarking in audio. The proposed system replaces redundant and imperceptible frequency components with hidden data. A psychoacoustic model is used to identify suitable frequency locations for data hiding. Such a method is complicated by the windowing and overlapping requirements used for signal conditioning. The proposed system uses data flipping in place of windowing and incorporates a novel data detection scheme with adaptive weighting to increase the robustness of the watermark transmission. The resistance of the watermarking system to filtering, amplitude scaling and additive white noise is measured and results presented.     

Synchronization for IR‐UWB System Using a Switching Phase Detector‐Based Impulse Phase‐Locked Loop

Conventional synchronization algorithms for impulse radio require high‐speed sampling and a precise local clock. Here, a phase‐locked loop (PLL) scheme is introduced to acquire and track periodical impulses. The proposed impulse PLL (iPLL) is analyzed under an ideal Gaussian noise channel and multipath environment. The timing synchronization can be recovered directly from the locked frequency and phase. To make full use of the high harmonics of the received impulses efficiently in synchronization, the switching phase detector is applied in iPLL. It is capable of obtaining higher loop gain without a rise in timing errors. In different environments, simulations verify our analysis and show about one‐tenth of the root mean square errors of conventional impulse synchronizations. The developed iPLL prototype applied in a high‐speed ultra‐wideband transceiver shows its feasibility, low complexity, and high precision.     

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.     

Kalman filter‐based channel estimation and ICI suppression for high‐mobility OFDM systems

The use of orthogonal frequency division multiplexing (OFDM) in frequency‐selective fading environments has been well explored. However, OFDM is more prone to time‐selective fading compared with single‐carrier systems. Rapid time variations destroy the subcarrier orthogonality and introduce inter‐carrier interference (ICI). Besides this, obtaining reliable channel estimates for receiver equalization is a non‐trivial task in rapidly fading systems. Our work addresses the problem of channel estimation and ICI suppression by viewing the system as a state‐space model. The Kalman filter is employed to estimate the channel; this is followed by a time‐domain ICI mitigation filter that maximizes the signal‐to‐interference plus noise ratio (SINR) at the receiver. This method is seen to provide good estimation performance apart from significant SINR gain with low training overhead. Suitable bounds on the performance of the system are described; bit error rate (BER) performance over a time‐invariant Rayleigh fading channel serves as the lower bound, whereas BER performance over a doubly selective system with ICI as the dominant impairment provides the upper bound. Copyright © 2008 John Wiley & Sons, Ltd.