Local echo canceler with optimal input for true full-duplex speechscrambling system

The purpose of this paper is to focus on the local echo-canceling problem of full-duplex scrambled speech communications over a two-wire telephone network when the scrambling transformation is located between the handset and body of a telephone. Such a design makes possible very efficient protection against electromagnetic compromising emanation, which in turn substantially enhances the overall security of a protected communication. We propose a new adaptive FIR filter algorithm for local echo cancellation in such applications. The proposed algorithm differs from the conventional one by the construction of input signals in an optimal way using the D-optimal experiment design. In this way, at each step, we generate a new sample of the D-optimal pilot sequence for the filter parameter estimation. Consequently, the adaptation of the local echo canceler is defined as an initialization process in the first phase of each protected telephone call. The advantage in using the proposed adaptive FIR echo canceler is demonstrated through simulation results     

A New Digital Echo Canceler for Two-Wire Full-Duplex Data Transmission

A new approach to echo canceling for two-wire fullduplex data transmission is proposed. The canceling signal is directly synthesized from the binary data, using a transversal filter approach, and the usual multiplications are replaced by additions and subtractions, thus allowing efficient operation of a large number of taps as required for the canceling of distant echoes. As a specific application, a system processing one sample per baud is discussed where timing signals at both communicating stations are assumed to be synchronized. A stochastic adjustment gradient-type algorithm is used for both training and adaptive tracking of the canceler. It is shown that convergence does not depend on intersymbol interference, timing phase, carrier phase, or the energy ratio of the local to the received signal, but is a function only of the number of taps. Convergence time is proportional to that number, and the optimum step size for fastest convergence is equal to the reciprocal of the number of taps. The residual fluctuation noise is proportional to that part of the mean-square (MS) error which cannot be reduced by the canceler and is a simple function of the product of the tap signal and the step size. The predicted convergence properties are verified by simulation results. Finally, it is shown how such an echo canceler might be used to allow two-wire full-duplex transmission for data rates as high as 4800 bit/s.     

Echo Canceler with Two Echo Path Models

An adaptive echo canceler with two echo path models is proposed to overcome the false adaptation problem for double-talking. The echo canceler possesses two separate echo path models (EPMs), one (background EPM) for adaptively identifying echo path transfer characteristics and the other (foreground EPM) for synthesizing an echo replica to cancel out echo. The parameter values of the foreground EPM are refreshed by those of the background EPM, according to a transfer control logic, when the logic determines that the background EPM is giving a better approximation of echo path transfer characteristics than the foreground EPM. Completely digital hardware implementation is described. Using the hardware, it is shown that virtually complete double-talking protection is actually realizable by the new method.     

Adaptive Reference Echo Cancellation

The AREC (adaptive reference echo cancellation) algorithm is presented for an echo canceler used in full-duplex two-wire digital transmission on digital subscriber loops. The AREC algorithm incorporates a decision-directed estimation of and compensation for the far-end signal which is a source of interference to the conventional echo canceler adaptation algorithm. The AREC algorithm thus offers much faster convergence and shorter coefficient wordlengths than the conventional algorithm. Analysis and simulation of the performance and convergence of both AREC and conventional echo canceler adaptation algorithms are carried out. Included in the analysis is the effect of receiver delay and coefficient wordlength requirements. A simple and robust startup procedure is proposed and investigated by simulation.     

Fast initialization of data-driven Nyquist in-band echo cancellers

A echo canceller fast training scheme for data-driven Nyquist in-band echo cancellers is presented. This scheme simultaneously estimates the desired near and far echo canceller coefficients by sending a special periodic training sequence and correlating a segment of the sequence with the real echo samples. The requirements and the generation of the training sequence are discussed. It is shown that the fast training method can also provide the parameters for the fast initialization of the phase roll compensator in the echo canceller, and thus complete the fast training of the entire echo canceller. Compared to the conventional LMS training and other training methods, this scheme provides accurate estimates of the echo canceller coefficients in a significantly reduced training time     

感觉加权回波消除器的研究

传统的自适应回波消除算法都是基于客观优化准则,而没有考虑回波消除的主观质量。本文提出在回波消除器中采用误差频率加权自适应滤波器结构,以充分利用人耳的听觉特性,提高回波消除的主观质量。客观测试和主观测试的仿真结果验证了新算法的有效性。     

Echo cancellation in DMT-receivers: circulant decomposition canceler

Asymmetric digital subscriber lines (ADSLs) employ discrete multitone modulation (DMT) as transmission format, where subcarriers are assigned to the up- and/or downstream transmission direction. To separate up- and downstream signals, the ADSL standard allows the use of echo cancellation resulting in improved bit rates, reach, and/or noise margins. In DMT-based modems, typically, the mixed time/frequency (MTF) domain echo canceling scheme, as proposed by Ho et al., is implemented. This technique estimates the echo filter in the frequency domain using the least mean square (LMS) algorithm with the transmitted echo symbols as update directions. Since not every tone of the transmitted echo signal will carry data, i.e., will be excited, the MTF adaptation process does not lead to a good estimate for the echo channel, unless extra power on unused echo tones is transmitted. However, transmitting extra power on such tones is often undesired. In this paper, we present an alternative echo canceling scheme referred to as the circulant decomposition canceler (CDC), which works without extra power requirements and with comparable complexity as the method of Ho et al. Similar to MTF echo canceling, the CDC scheme can easily be incorporated into a multirate environment with different transmit and receive rates and can also cheaply be combined with per-tone equalization and double talk cancellation to allow fast tracking and/or convergence in the presence of a far-end signal.     

A multirate acoustic echo canceler structure

A new subband echo canceler (SBEC) structure is proposed to reduce the transmission delay introduced by conventional SBEC structures, without distorting the near-end signal. The proposed structure is based on computing two output errors, one for using during single-talk and the other one for using during double-talk periods. With the SBEC structure we propose a double-talk detector with a subband configuration which allows a fast and accurate detection of double-talk periods, enabling the SBEC algorithm to track changes in the echo path impulse response when the near-end signal is absent. Computer simulations using actual speech signals, and subjective evaluation tests are given to show the convergence performance, tracking and double-talk detection ability, of the proposed scheme     

Improvement in adaptive noise canceling using a nonlinear filter based on the pontryagin minimum principle

This paper introduces a new nonlinear filter that is used for adaptive noise canceling. The derivation and convergence properties of the filter are presented. The performance, as measured by the root mean square error between the signal and its estimate, is compared with that of the commonly used least mean square (LMS) algorithm. It is shown, through simulation, that the proposed nonlinear noise canceler has, on the average, better performance than the LMS canceler. The proposed adaptive noise canceler is based on the Pontryagin minimum principle and the method of invariant imbedding. The computational time for the proposed method is about 10% of that of the LMS, in the studied cases, which is a substantial improvement.     

A stereo echo canceler with correct echo-path identification basedon an input-sliding technique

A new stereo echo canceler with correct echo-path identification based on an input-sliding technique is proposed. A time-varying filter located in one of the two channels periodically delays the input signal. By this input sliding, the correct echo-path identification is achieved. Aliasing components and audible clicks by input-sliding are made inaudible by selecting appropriate parameter values for the time-varying filter. Simulations with the NLMS algorithm and a white Gaussian signal confirm the correct echo-path identification. The subjective quality of the input signal with slides is 4.38 based on the ITU-R five-grade impairment scale. Experimental results based on an implementation by 32-bit floating-point digital signal processors show that ERLE is not degraded by talker changes in the remote room. The mean opinion score is as much as 0.55-point higher than the conventional stereo echo canceler for different round-trip delays     

A new configuration for echo canceller adaptable during double talkperiods

A three-port echo canceler (EC) configuration is proposed which observes the signal of the near-end side on a two-wire circuit in addition to the four-wire circuit signals. Embedding these signals on hybrid ports into a three-dimensional autoregressive process, echo path and innovations of near- and far-end speeches can be estimated through a three-channel lattice filter. The new configuration is then able to track echo path time variance, even during double talk (DT), and requires no changeover at either the beginning or end of DT, thus eliminating the need for DT detection. Two echo synthesizers utilizing inverse lattice and the echo path estimate possess guaranteed stability without the need for testing     

A New Digital Echo Canceler for Two-Wire Subscriber Lines

A new type of digital echo canceler for two-wire digital transmission is presented. The new principle involves very simple signal processing and is thus an interesting alternative for digital transmission on subscriber lines. The principle is compared with other echo cancellation techniques, and it is shown how choice of line code, equalization, and carrier recovery are affected by the new echo canceler. A theoretical analysis of the principle is given, taking into account finite accuracy, jitter, noise, and correlated data streams. The echo canceler can be used for line attenuation up to 40 dB. At 80 kbits/s this corresponds to at least 7 km 0.6 mm cable and is sufficient to cover more than 99 percent of the existing Norwegian subscriber lines.     

Fast initialization of equalizers for VSB-based DTV transceivers in multipath channel

This paper proposes a fast initialization technique for equalization of 8-VSB-based digital television (DTV) signal in severe multipath channels. We consider the use of a modified decision feedback equalizer (MDFE) , for fast initialization. The feedback filter (FBF) of the MDFE can be initialized simply by estimating the channel impulse response and only the feedforward filter (FFF) of the MDFE need training for initialization. To overcome the shortage of the training sequence in the VSB DTV signal, we propose a new initialization method by generating a virtual training signal to initialize the FFF of the MDFE. Simulation results show that the proposed scheme can fast initialize the equalizer using less than 5000 symbols, while providing the receiver performance comparable to that of conventional schemes.     

Acoustic echo-canceler using the FBAF algorithm

An acoustic echo-canceler for teleconferencing systems is realized based on the frequency bin adaptive filtering (FBAF) algorithm. In the FBAF algorithm, each frequency bin does an independent adaptive filtering, so that parallel processing can be used to increase the throughput of the system. Hardware size can be reduced to about 25% of the FIR time domain adaptive filter (TDAF) requirement. The realized echo canceler allows a comfortable conversation with only 8 ms of delay. The hardware prototype contains 12 VSP chips and one DSP chip, An ERLE (echo return loss enhancement) of 30 dB was achieved using this prototype hardware for an echo reverberation path with 260 ms delay. An efficient method for normalizing the convergence factor of the FBAF algorithm with a correlated input signal is given that speeds up the convergence rate. The performance is shown by computer simulation     

Echo cancellation in the telephone network

Most long-distance telephone connections generate echoes, which must be heavily attenuated in order to obtain satisfactory transmission quality. Voice-actuated switches (echo suppressors) are widely used to eliminate echoes but have an unfortunate tendency also to cut out part of the desired signal from the other end of the line. Because the distortion caused by echo suppressors is particularly noticeable on satellite-routed connections, the advent of telephone communication via satellite, including the recent introduction of satellite circuits into the U.S. domestic network, has motivated the search for a better way to eliminate echoes. The answer may be the echo canceler, an adaptive filter which selectively eliminates echoes. Advanced echo canceler designs have been undergoing field trials in recent years. This article explains why echo cancelers are advantageous and how they work.     

Timing Recovery in Digital Subscriber Loops Using Baud-Rate Sampling

Sampled-data techniques are the most practical means of obtaining the necessary signal processing functions for timing recovery in the VLSI implementation of a digital subscriber loop transceiver. The sampled-data timing recovery techniques described in this paper are applicable to both echo cancellation and time-compression multiplexing systems. Timing recovery using baud-rate sampling in conjunction with a special pulse-shaping and timing function fulfills all the objectives for timing recovery in this application. It recovers a timing phase that has minimum precursor intersymbol interference, and makes possible the combination of decision feedback equalizer and echo canceler, reducing the convergence time and increasing the step size. The pulse-shaping function can be performed either in the transmitter by means of digital coding, or in the receiver by means of analog filtering. In the latter case, the transmitted pulse is compatible with more conventional approaches. The proposed partial-response line coding, a special form of AMI coding, is less susceptible to line impairments if detected as a two-level signal. Performance by analysis, simulation, and experimental measurements is reported on a variety of cable configurations, some including bridged taps. Analysis of jitter performance leads to design techniques for reducing the jitter magnitude.     

Adaptive harmonic jammer canceler

A new adaptive harmonic jammer canceler is proposed. It is based on the use of two sensors that enable an adaptive generation of a reference signal that is uncorrelated with the desired signal. This reference signal is used for the reconstruction of the desired signal by an adaptive subtraction method. This canceler is well suited to radio communications. A theoretical analysis of the convergence of the coupled algorithms is presented with the help of the associated ordinary differential equation introduced by L. Ljung. Numerical simulations illustrate the different proposed algorithms     

A new double-talk detection algorithm based on the orthogonalitytheorem

The algorithm not only prevents the echo canceler from being disturbed by double talking but also tracks the echo path variations. Although the algorithm requires more computation and storage than conventional algorithms, excellent double-talk interference protection performance and echo path tracking have been obtained     

Asynchronous and Timing Jitter Insensitive Data Echo Cancellation

An approach to the implementation of asynchronous and timing jitter insensitive data echo cancellation is described. This approach introduces a small amount of jitter in the transmitted data signal, or alternatively in the received signal sampling, and uses a simple digital phase-locked loop together with the storage of two sets of echo canceler coefficients. The effect of derived timing jitter on the echo cancellation accuracy is completely eliminated for a loop timed transceiver (as in a digital subscriber loop network termination transceiver), and is easily reduced to negligible levels for a nonloop timed transceiver (as in a digital subscriber loop line card transceiver or a voiceband data modem). In the case of a voiceband data modem, this approach is one method to achieve asynchronous echo cancellation without the need to recover and resample a continuous-time far-end data signal.     

A new IIR adaptive echo canceler: GIVE

A novel IIR adaptive gradient instrumental variable echo canceler (GIVE) is presented. Its features include adaptive controllability during double-talk periods in acoustic conference systems; guarantee of global convergence; low computational cost (the same order as the IIR LMS algorithm of the equation error method); and flexible structures (parallel or series-parallel structures). We also show a convergence analysis for gradient adaptive algorithms including GIVE. Based on this analysis, the optimum stepsize for GIVE and three suboptimum algorithms are proposed to accelerate convergence and reduce misadjustment. In addition, a simple method that guarantees the stability of IIR filters and a configuration of GIVE applicable to closed loop systems are presented. These proposals are extensively studied by computer simulations