共查询到20条相似文献,搜索用时 15 毫秒
1.
The authors present a novel algorithm for echo cancellation. The algorithm consists of simultaneously applying the LMS algorithm to the near-end section of the echo canceller, and a controlled mixed LMS-LMF algorithm to the far-end section. This combination results in a substantial improvement in performance of the proposed scheme over the LMS and the LMF algorithms 相似文献
2.
Proportionate adaptive algorithms for network echo cancellation 总被引:2,自引:0,他引:2
By analyzing the coefficient adaptation process of the steepest descent algorithm, the condition under which the fastest overall convergence will be achieved is obtained and the way to calculate optimal step-size control factors to satisfy that condition is formulated. Motivated by the results and using the stochastic approximation paradigm, the /spl mu/-law PNLMS (MPNLMS) algorithm is proposed to keep, in contrast to the proportionate normalized least-mean-square (PNLMS) algorithm, the fast initial convergence during the whole adaptation process in the case of sparse echo path identification. Modifications of the MPNLMS algorithm are proposed to lower the computational complexity. 相似文献
3.
A single chip, 128 coefficient, asynchronous echo canceller is presented. Cancellation is performed by an FIR filter whose coefficients are adapted using the power-of-two modified LMS algorithm. The pipelined circuit updates all coefficients and generates the filtered output every cycle while allowing a sampling rate >206.5 kHz 相似文献
4.
Hybrid LMS-LMF algorithm for adaptive echo cancellation 总被引:1,自引:0,他引:1
Zerguine A. Bettayeb M. Cowan C.F.N. 《Vision, Image and Signal Processing, IEE Proceedings -》1999,146(4):173-180
The coefficients of an echo canceller with a near-end section and a far-end section are usually updated with the same updating scheme, such as the LMS algorithm. A novel scheme is proposed for echo cancellation that is based on the minimisation of two different cost functions, i.e. one for the near-end section and a different one for the far-end section. The approach considered leads to a substantial improvement in performance over the LMS algorithm when it is applied to both sections of the echo canceller. The convergence properties of the algorithm are derived. The proposed scheme is also shown to be robust to noise variations. Simulation results confirm the superior performance of the new algorithm 相似文献
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In this paper an algorithm is presented for adaptive filtering in the frequency-domain with application to acoustic echo cancellation. This algorithm, called generalized multi-delay filter(GMDFα, is derived from the frequency-domain implementation of the time-domain block least mean square algorithm. Two different implementations are introduced, one based on the discrete Fourier transform(dft) and one based on the discrete Hartley transform(DHT); some results on fixed-point implementation of the algorithm are provided which are compared to results obtained from floating point implementation. Finally, the application of thegmdfα algorithm to acoustic echo cancellation, in hand-free telephone systems, is detailed. Some control strategies are presented; in particular a novel double-talk detector based on a spectral dissimilarity measure is introduced ; also, a twin-filter structure which significantly enhances the echo rejection is derived. 相似文献
8.
Qu Jin Zhi-Quan Luo Kon Max Wong 《Signal Processing, IEEE Transactions on》1996,44(7):1669-1680
The application of multirate filter banks in echo cancellation is investigated. The multiresolution algorithm is used to decompose the received sampling sequence into a number of components, and then, an adaptive algorithm is applied to cancel the echo in the received signal. In this paper, the performance of this method is discussed, from which optimal conditions for echo cancellation are established for the design of wavelet packet multiresolution decomposition. An efficient algorithm for designing such a set of optimal discrete filter banks is developed. The cases of optimal in-band and adjacent-band adaptive filtering are examined. Experimental results showed that the use of optimally designed multiresolution filter banks coupled with in-band or adjacent-band adaptive filtering is much more effective than the employment of commonly used wavelet filter banks. Furthermore, the use of the adjacent-band adaptive filtering algorithm has superior performance compared with that of the in-band filtering 相似文献
9.
Chin W.H. Farhang-Boroujeny B. 《Vision, Image and Signal Processing, IEE Proceedings -》2001,148(4):283-288
Adaptive filtering in subbands was originally proposed to overcome the limitations of conventional least-mean-square (LMS) algorithms. In general, subband adaptive filters offer computational savings, as well as faster convergence over the conventional LMS algorithm. However, improvements to current subband adaptive filters could be further enhanced by a more elegant choice of their design/structure. Classical subband adaptive filters employ DFT-based analysis and synthesis filter banks which results in subband signals that are complex-valued. The authors modify the structure of subband adaptive filters by using single-sideband (SSB) modulated analysis and synthesis filter banks, which result in subband signals that are real-valued. This simplifies the realisation of subband adaptive filters 相似文献
10.
Hadi Sadoghi Yazdi Masoud Rezaei 《Analog Integrated Circuits and Signal Processing》2010,64(2):191-198
In this paper, a new analog adaptive filter is introduced with application in adaptive echo cancellation namely, the Wheatstone
bridge-based analog adaptive filter (WAAF). It is proved the WAAF is a variable weight analog IIR filter. IIR filter weights
vary with gate-source voltage control of a MOSFET transistor in triode region. The best balance point control of the WAAF
is achieved using least mean square (LMS) algorithm. It is proved that analog LMS algorithm converges faster than digital
LMS adaptive filter. The superiority of the proposed WAAF is observed in the designing process, computational cost, convergence
speed and real time operation. Also, experimental results show ability of the proposed WAAF in the hybrid circuit of the telephone
echo cancellation. 相似文献
11.
针对通信中的回波问题,基于自适应滤波的最小均方(LMS)算法,设计了一个自适应回波抵消器。基于FPGA芯片,在DSP Builder平台上,有效结合MATLAB/Simulink和Quartus Ⅱ设计工具,根据模块化设计思想实现了LMS算法自适应回波抵消器硬件电路设计。软件仿真和系统FPGA硬件实测结果表明,该设计方法使回波抵消器的FPGA硬件实现更加简便快捷。 相似文献
12.
Cristian Contan Botond Sandor Kirei Marina Dana Topa 《Signal, Image and Video Processing》2016,10(3):511-518
This paper proposes novel acoustic echo cancellation (AEC) approaches based on linear and Volterra structures. The AECs use modified normalized least-mean-square (NLMS) updates to improve the convergence and to maintain the same steady-state misadjustment. In the first case, starting from a new cost function, the resulting variable step size depends on the instant error value and on an estimated error threshold. Secondly, the need of beforehand steady-state error threshold estimation is removed by an automatic step-size control involving the absolute error envelope evolution. The methods are tested for an acoustic enclosure setup modeled using measured linear and quadratic kernels, and their behavior is compared to that of the traditional NLMS and another technique found in the open literature. Also, they are tested for a change in the echo path and for assorted nonlinearity and local signal powers. The comparison is made in terms of the echo-return loss enhancement for WGN and speech as excitation. The simulations show that the proposed adaptations offer increased convergence rates for the same steady-state error. 相似文献
13.
A new echo cancellation structure for discrete multitone systems is presented, where each used tone has its own per-tone echo canceller in addition to a per-tone equalizer, which provides an alternative to currently employed time domain and time/frequency domain approaches. The per-tone approach enables us to optimize the signal-to-noise ratio for each tone separately by solving a minimum mean-square error problem for each tone, with implicit so-called joint shortening. Complexity during data transmission is compared for time domain, time/frequency domain, and per-tone echo cancellation. Structures with reduced complexity are derived for an interpolated and a decimated rate setup. Finally, simulation results for an asymmetric digital subscriber line setting demonstrate improved performance over time domain (or time/frequency domain) echo cancellation. 相似文献
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Teleconferencing systems and hands-free mobile terminals use acoustic echo cancellation (AEC) for high-quality full-duplex speech communication. The problem of aliasing in subband AEC is addressed. Filter banks with implicit notch filtering are derived from cascaded power symmetric-infinite impulse response (CFS-IIR) filters. It is shown that adaptive filters used with these filter banks must be coupled via continuity constraints to reduce the aliasing in the residual echo. A continuity constrained NLMS algorithm is therefore proposed and evaluated 相似文献
16.
The history of echo cancellation 总被引:1,自引:0,他引:1
《Signal Processing Magazine, IEEE》2006,23(5):95-102
17.
一种新NLMS自适应滤波算法及其在多路回波消除中的应用 总被引:3,自引:0,他引:3
提出一种NLMS改进算法并对其收敛性进行了证明。该算法计算复杂度低于Sankaran(1997)所提出的带有正交改正因子的归一化算法(NLMS-OCF)和仿射投影算法(APA),并具有易于实现等特点。仿真结果表明,以单路语音信号作输入时,新算法具有比NLMS-OCF算法更好的收敛速度和精度,而在收敛速度和精度相当的情况下,新算法比APA算法所占用的CPU时间少。将新算法扩展成两路算法后,扩展算法仍然保持了这些特点,与Sankaran(1999)两路NLMS-OCF及Benesty(1996)所提多路仿射算法(APA-MC)相比,新算法更适合于应用到多路回波消除等实时性要求高的场合。1 相似文献
18.
The problem of echo cancellation in a multitone modulation (MTM) scheme is addressed. A general model for the near-end echo is derived and is used to identify candidate data-driven echo canceller (DDEC) structures. The stability, steady state performance, and associated system complexity of an adaptive DDEC based on the stochastic gradient approach is developed. It is further shown how the symmetry in the derived echo path model can be exploited to enable a reduction in canceller complexity, enhancing convergence speed without sacrificing final SNR. Simulation results are provided that confirm the analytical predictions 相似文献
19.
A method for designing near optimal, tapered subarrays for adaptive interference cancellation is proposed. The design method simultaneously produces a complete ordered set of fixed beam definitions, or nonadaptive weight vectors. The designer may choose to implement the first K of these if he or she wishes to have exactly K adaptive weights. In other words, the digital-adaptive processing is done in beam space, such that the beams are designed using the proposed method. To facilitate an RF implementation of the nonadaptive beamformer, each auxiliary beam uses only a designer-specified number of the elements in the aperture, thereby reducing the number of waveguide connections required. This design approach is fundamentally different from conventional subarray design approaches in that the new designs utilize cost functions related to interference cancellation. 相似文献
20.
In acoustic echo cancellation (AEC), the sparseness of impulse responses can vary over time or/and context. For such scenario, the proportionate normalized subband adaptive filter (PNSAF) and μ-law (MPNSAF) algorithms suffer from performance deterioration. To this end, we propose their sparseness-measured versions by incorporating the estimated sparseness into the PNSAF and MPNSAF algorithms, respectively, which can adapt to the sparseness variation of impulse responses. In addition, based on the energy conservation argument, we provide a unified formula to predict the steady-state mean-square performance of any PNSAF algorithm, which is also supported by simulations. Simulation results in AEC have shown that the proposed algorithms not only exhibit faster convergence rate than their competitors in sparse, quasi-sparse and dispersive environments, but also are robust to the variation in the sparseness of impulse responses. 相似文献