Line Echo Cancellation Research Articles

Fractional LMS and NLMS Algorithms for Line Echo Cancellation

In long haul communication environments, speech data transmission is severely affected by echoes. This phenomenon results in high bit errors as well as in degraded and annoying performance. Traditionally these problems, including hybrid and acoustic echoes, have been controlled through the use of echo suppressors. These suppressors were subsequently replaced by line echo cancellers using adaptive Finite Impulse Response filters. Fractional calculus has been applied successfully for fixed filtering with constant coefficients and in discrete time adaptive filtering that adjusts the weights according to the environment. This paper presents the Fractional Least Mean Square (FLMS) and Fractional Normalized LMS (FNLMS) algorithms for application in echo cancellation. Moreover, the performances of the FLMS and FNLMS are compared with those provided by the standard LMS, NLMS and Block Discrete Fourier Transform solutions. The mean square error criterion is used as the performance comparison criterion for two types of voice signals namely real and synthetic. The simulation results show a performance improvement of about 50% over the traditional counterparts.

A Robust and Adaptive Line Echo Canceller

In the actual communication system, the line echo is a frequent problem, so the line echo canceller is necessary. Traditionally, the cancellers are designed according to line noises based on the Gaussian noise model. However, the noises do not obey the Gaussian distribution; therefore the performances of algorithms decrease, even cannot be used. In order to solve this problem, this paper proposes a line echo canceller based on alpha stable distribution model. The Method in this line echo canceller has Robust and Adaptive. Alpha stable distribution modeling impulse noise, and under the rule of minimum average <i>p</i> norm, adaptive time delay estimator offsets the echo. This method can not only effectively offset the line echo with the Gaussian noise, but also to have the effect of good to echo with impulse noise. Theoretical analysis and computer simulation experiments show that it is very kind of the line echo canceller. It is robust to impulse noises and adaptive to echo cancellation.

The Dynamic LMS for Line Echo Cancellation

To the best of our knowledge, this study represents the proposal using the dynamic least mean square (DLMS) algorithm to reduce the computation load of LMS. Moreover, three regions of impulse response of line echo path are also proposed to analyze the redundant coefficients. Using the DLMS method, redundant coefficients can be detected and grouped, thereby automatically reducing computation. We employed line echo cancellation (LEC) to evaluate the performance of DLMS. The pure-delay and overlong regions of impulse response of line echo path are grouped and the associated computation load is reduced. The experimental results confirm the excellent performance of DLMS achieving a 35% savings in computation. Moreover, the quality echo return loss enhancement (ERLE) of DLMS also maintains at a level nearly equal to LMS.

Learning Echo Paths During Continuous Double-Talk Using Semi-Blind Source Separation

Echo cancelers typically employ control mechanisms to prevent adaptive filter updates during double-talk events. By contrast, this paper exploits the information contained in time-varying second order statistics of nonstationary signals to update adaptive filters and learn echo path responses during double-talk. First, a framework is presented for describing mixing and blind separation of independent groups of signals. Then several echo cancellation problems are cast in this framework, including the problem of simultaneous acoustic and line echo cancellation as encountered in speaker phones. A maximum-likelihood approach is taken to estimate both the unknown signal statistics as well as echo canceling filters. When applied to speech signals, the techniques developed in this paper typically achieved between 30 and 40 dB of echo return loss enhancement (ERLE) during continuous double-talking.

Carrier Class High Density VoIP Media Gateway using Hardware Software Distributed Architecture

This paper presents a system on chip (SOC) based high-density carrier class Voice over Internet Protocol (VoIP) media gateway. The design exploits three levels of parallelism namely HW/SWpartitioning, multiprocessing and instruction level parallelism. The system facilities the carrier class service providers to transform channels on PSTN network to IP network, enabling the consumers to enjoy several value added services, such as voice, data, fax and video all through a uniform network at lower cost. The SoC implements a high-density media gateway switch for carrier class VoIP applications. The solution is synthesized to operate at 266 MHz handling up to OC-3 data rate port density equivalent to 2016 simultaneous calls on a single chip and it can be cascaded to support OC-12 and OC-48 rate port density on single board in different configurations. On each channel the SoC supports up to 128 ms tail length G.168 compliant Line Echo Cancellation (LEC), voice activity detection (VAD), comfort noise generation (CNG) and discontinuous transmission (DTX), dual tone multi frequency (DTMF) detection and generation, variety of G.xxx voice codecs, V. 17 Fax and V.34 data modem standards and MPEG4 video compression/decompression algorithms. The solution is compared with the existing architectures and commercial products; it offers 3x to 10x cost, size, power and port density performance improvements with IP and PSTN internetworking support. The media gateway system based on the presented architecture has been used to develop IP-based automatic call distribution (ACD) by etel Pvt. Ltd and installed in Telephone Inquiry Exchange of Pakistan Telecom and its variants are being tested by Buraq Telecom for installation in their service network.

A Method for Reduced Finite Precision Effects in Parallel Filtering Echo Cancellation

<para xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> The two-path algorithm is an adaptive filter algorithm based on a parallel filter structure, which has been found to be useful for line echo cancellation as well as for acoustic echo cancellation. It is well known that in finite precision arithmetic, the adaptation process of adaptive algorithms can be reduced or even halted due to finite precision effects. This paper proposes a variant of the two-path scheme where the effects of quantization are reduced, without any significant increase in complexity. The improvement is shown by simulations using bandlimited flat spectrum noise as well as real speech signals. </para>

Sub-band Implementation of Adaptive Nonlinear Filter for Adaptive Nonlinear Echo Cancellation

The adaptive Volterra filter has been successfully applied in nonlinear acoustic echo cancellation (AEC) systems and nonlinear line echo cancellation systems, but its applications are limited by its required computational complexity and slow convergence rate, especially for systems with long memory length. In this paper, we first apply a more general nonlinear filter, the function expansion nonlinear filter, in the acoustic echo cancellation - the Volterra filter can be regarded as special case of the function expansion nonlinear filter. Then by leveraging to a multi-channel configuration of the function expansion nonlinear filter and the sampling theory for nonlinear systems, we extend linear sub-band delay-less adaptive filter techniques to develop an efficient sub-band implementation of the adaptive function expansion nonlinear filter. The developed sub-band configuration of the adaptive nonlinear filter can greatly improve the convergence rate and reduce the computational complexity of nonlinear echo cancellers, which is shown by analyses and simulations.

Measurement of the Effects of Temporal Clipping on Speech Quality

This paper investigates the effects of temporal clipping on perceived speech quality. Temporal clipping usually results from voice activity detection (VAD), or line echo canceller's nonlinear processor, and the clipped speech portions are replaced by comfort noise. A nonintrusive algorithm is proposed to predict speech quality based on the clipping statistics. Mean opinion score (MOS) is used as a metric for speech quality and is measured by perceptual evaluation of speech quality (PESQ). The impacts of speech frame size and noise spectrum on the algorithm are also investigated. The results show that the proposed algorithm can efficiently predict the speech quality. The correlation coefficient between the prediction and the measurement is about 0.975, and the root mean square error for the prediction is 0.20 MOS. The algorithm can be used as an integral part of a general speech quality assessment scheme in voice over Internet protocol (VoIP)

Adaptive line echo cancellation combined withCVSD speech coders

An efficient method for controlling the echo in a continuously variable slope delta modulation (CVSD) mobile communication channel is proposed. By utilising the symmetry of the coder pairs, the proposed method shortens the echo path length and reduces the nonlinear distortion inherent in the coders. An additional advantage for the VLSI implementation is obtained using the combined bit-stream. The proposed method shows an improved echo return loss enhancement (ERLE) performance and reduced complexity compared to the conventional method.

Asymmetric digital subscriber line echo cancellation

To increase the convergence speed and reduce computational complexities associated with an echo canceller used to cancel digital subscriber loop echo responses with long tails, a new echo canceller is proposed which is composed of a cascade of a lattice and an orthogonalised infinite impulse response (IIR) (CALAOI) filter. The propose echo canceller was found to have a higher convergence speed and substantially reduced complexity compare with those of a conventional finite impulse response (FIR) echo canceller.

Performance evaluation of a DSP-based teleconferencing system

Simulation and real time results associated with acoustic and line echo cancellation for a full-duplex teleconferencing system are presented. Both broadband (conventional) and subband finite impulse response (FIR) adaptive acoustic echo cancellers are considered. In subband scheme, quadrature mirror filters (QMF) are employed in order to construct an eight subband acoustic echo canceler. The normalized mean square (NLMS) is employed for updation of adaptive filter coefficients. Real time digital signal processing is conducted on a floating-point DSP chip in order to assess the dB cancellation of acoustic echo and line echo cancellers.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

A finite precision analysis of the block-gradient adaptive data-driven echo canceller

The block-gradient (block LMS) algorithm's finite precision performance in the data-driven echo canceller application is analyzed. From both the analysis and simulation results, it was found that the block LMS requires significantly less precision than the standard LMS algorithm. The analysis also shows how higher convergence and tracking speeds may be traded for an improvement in precision requirements. The authors derive formulae that can be used to accurately compute echo rejection levels as a function of precision, signal powers, step-size, block length, and echo canceller length. The utility of the formulae is demonstrated by showing performance levels for a typical V.32bis full-duplex voiceband modem operating at a transmission rate of 14.4 kbit/s (7200 Hz sampling rate), and for a high-speed digital subscriber line echo canceller operating at a sampling rate of 400 kHz. >

High bit rate digital subscriber line echo cancellation

The authors explore the issues of complexity, precision, and dynamic range and practical implementation issues such as nonlinearities in high bit rate digital subscriber line (HDSL) echo cancelers. The approach taken is to study these issues analytically using computer simulations, and then to verify simulation results with laboratory prototyping. One of the objectives was to determine whether HDSL echo cancellation will be practical and cost effective within the next several years, consistent with the desire to develop and deploy a repeaterless T1 capability by 1993. Thus, attention is given to practical design issues.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Acoustic echo cancellation in subbands

Acoustic echo cancellation for teleconferencing or hands‐free telephony constitutes a challenging task for today's digital signal processing techniques. Compared to the well‐established line echo cancellation in telephone networks, the problem takes on a considerably larger size: The impulse response of the echo path to be compensated is longer and may vary more rapidly in time. Moreover, larger bandwidths are often desirable. Therefore, acoustic echo cancellation requires algorithms that adapt far more coeffecients and converge faster than those commonly used in line echo cancelers. On the other hand, direct implementation of sufficiently advanced adaptive filtering algorithms is usually prohibited because of their computational complexity and inherent numerical difficulties. As a way out, the subband approach realizes a “divide and conquer” strategy and promises a reduction of computational complexity and favorable circumstances for fast convergence at the cost of some extra delay. After comparing several possible subband structures, the frequency subband structure is discussed in some detail. For adaptation using LMS‐type algorithms, reasons for the improved convergence behavior compared to a fullband implementation are explained. Design examples are presented to illustrate the effectiveness of the method.