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1.
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.  相似文献   

2.
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.  相似文献   

3.
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.  相似文献   

4.
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.  相似文献   

5.
For full-duplex high-speed data transmission over the two-wire line using the same frequency band, it is required to sufficiently suppress the echo. The use of a conventional adaptation method may take a long time to train the echo canceler. Fast training can be achieved by initializing the coefficients of the echo canceler with an estimate of the impulse response of the echo path. We propose a method for fast initialization of the echo canceler by using a circular convolution technique. The proposed method enables the use of real-valued training signals instead of complex-valued ones, resulting in significant reduction of the initialization time as well as the implementation complexity. Finally, the performance of the proposed method is analyzed and verified by computer simulation.  相似文献   

6.
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.  相似文献   

7.
Adaptive mean-square-tapped-delay-line echo cancellers for voice applications are conventionally designed to stop adjustment during periods of "double-talking", i.e., when a large informationbearing signal is present along with the echo signal to be cancelled. Continuous adaption is, however, desirable in full-duplex, two-wire data transmission where the periods of double-talking are so long that the echo channel may vary. We presume that the tap weights of an echo canceller have converged during a training period free of double talking, and address the problem of subsequent echo-canceller tap adjustment via the estimated-gradient algorithm in the presence of double talking. In the estimated-gradient algorithm the tap increment should be proportional to the product of the residual echo and the tap voltage. However, when double talking occurs the residual echo can only be estimated. For an idealized double-talking model, it is demonstrated, from infinite-precision considerations, that use of the memoryless maximum-likelihood estimate of the residual echo is nearly equivalent to abrupt reduction of the step size of the adjustment algorithm when double-talking begins, and could provide an automatic mechanism for recognizing double-talking. Unfortunately, the response of a digitally implemented canceller to a sharply reduced step size can be a deterioration in performance. In fact, the use of an exceedingly small step size during periods of doubletalking may lead to a cancellation error considerably larger than that predicted by coefficient precision. It is demonstrated how averaging the estimated gradient can significantly decrease the mean-squared tap error during periods of double talking. To a first approximation, the tap-weight error can be reduced by a factor proportional to the averaging interval, with an equivalent decrease in tracking capability.  相似文献   

8.
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  相似文献   

9.
A digitally-implemented echo canceller operating at a rate greater than twice the highest passband frequency is proposed for fullduplex data transmission on a two-wire circuit. Located at each end of the communication circuit, the cancellers operate independently of the local receivers and do not require synchronization of the two data stations. Economies in storage and A/D conversion, in comparison with voice-type cancellers described in the literature, are achieved by accepting data symbols as reference input instead of samples of the transmitted line signal. Convergence of transversal filter tap weights is demonstrated under a mean-squared error criterion, and use of the real passband error rather than the complex analytic error is found to lead to the same residual error at the expense of a doubled convergence time. An operational protocol and adaptation algorithm are proposed which make possible both rapid start-up and slower adaptation during double talking. Provision is made for limiting the number and location of active taps on the transversal filter to those actually necessary for replicating the echo channel, resulting in two transversal filter sections of moderate length separated by a bulk delay. Results from a computer simulation of the proposed canceller are offered to demonstrate that convergence of mean-squared tap-weight error follows the predicted exponential characteristic, that the length of the bulk delay can be determined from a single channel sounding under typical channel noise conditions, and that the use of an averaged-gradient algorithm will allow the canceller to adapt, although slowly, to a change in the echo channel during full-duplex operation.  相似文献   

10.
在强相关大动态范围语音作用下,回声抵消器直接更新上千阶自适应滤波器系数,计算量大、收敛速度慢。采用子带分析与合成的方法能够减少计算量和提高算法的收敛性能,但子带的分析与合成也给线路中引入了信号延迟。提出了一种改进的双通道滤波回声抵消算法,将子带自适应滤波器映射到全频带滤波器,减少了信号的延迟;同时采用双通道滤波器。使系统工作在较小残余回声功率下。仿真结果表明,改进算法在单边会话情况下收敛快.具有较好的回声抵消效果。  相似文献   

11.
We analyze the effect of Subscriber-end timing recovery Circuit jitter on the performance of two types of adaptive echo cancellers that can be used for full-duplex digital transmission on two-wire subscriber loops. Under severe echo-to-far-end signal ratios, echo canceller performance is found to be quite sensitive to high-frequency jitter components. Satisfactory performance with respect to jitter requires that a narrow-band phase-locked loop, rather than a single-tuned high-Qfilter, be employed for timing recovery.  相似文献   

12.
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  相似文献   

13.
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.  相似文献   

14.
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  相似文献   

15.
Development of compact, high-speed, and low power adaptive transversal filters (ATF's) has been of great interest. Such ATF's have already been successfully fabricated in monolithic form employing a short channel MOSFET process and switched capacitor technique, using only complete analog circuit technology while eliminating inherent analog problems. This IC with five taps works with both ±5 V and 10 V supplies at a clock rate of more than 250 kHz. It has been designed for multipurpose applications such as a decision feedback equalizer (DFE), echo canceller (EC), and linear equalizer (LE), the last of which has already been reported [5]. The architecture to realize these applications will be described first. Much discussion is presented on the investigation of the operation, characteristics, and performance limitations of this IC in the DFE mode, from which those in the EC mode for two-wire full-duplex data transmission can be understood. For a single echo with magnitude of one half that of the original signal and delay time of one clock period from the original signal, and convergence factor of 0.2 for weight adaptation, the DFE operating at the clock rate of 100 kHz completes equalization within about 1.2 ms resulting in a residual rms distortion of -45.4 dB. The dominant performance limitation factors are found to be both the correlator harmonic distortion and transversal filter noise, but not the convolver harmonic distortion which is the dominant factor for the LE.  相似文献   

16.
This paper proposes an echo cancellation algorithm that can be used for full-duplex digital data transmission over existing twisted-pair cables. The proposed algorithm is a form of the "sign algorithm." A new adaptation method is introduced to change the canceller step size according to the residual echo level by using the correlation between the echo and the error signal. The adaptation method can increase the echo canceller convergence speed by an order of magnitude over the conventional sign algorithm. An echo canceller utilizing the proposed algorithm is suitable for VLSI implementation, because it requires no multibits analog-to-digital converter. Theoretical analysis and parameter optimization are carried out. The rapid convergence property of the algorithm is verified by computer simulation as well as experiments.  相似文献   

17.
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  相似文献   

18.
Consideration is given to the influence of the noise and data sequences present in the received data signal on a nondecision-aided timing recovery scheme in digital modem receivers, It is known that white noise is not particularly disturbing for timing recovery, whereas data signals such as local echo (or residual echo) introduce a bias, called jitter, into the sampling recovered phase. It is shown that when the disturbing data signal has power P less than the power S of the useful signal (whose timing must be recovered), the jitter is sinusoidal with amplitude proportional to the ratio P/S. In the opposite situation, the bias increases indefinitely with time  相似文献   

19.
This paper describes a new technique for implementing an echo canceller for full-duplex data transmission (such as in digital subscriber loops and volceband data sets). The canceller can operate in spite of time-invariant nonlinearities in the echo channel or in the implementation of the echo canceller itself (such as in the D/A converters). The basic structure of the linear echo canceller is not changed, but taps are simply added to account for the nonlinearity. The number of taps which must be added depends on the degree of nonlinearity which must be compensated. Numerical results based on computer simulation are given which show that typical nonlinearities encountered in MOS D/A converters can be compensated by a relatively small number of taps added to the linear echo canceller, and substantial improvement in the cancellation results.  相似文献   

20.
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.  相似文献   

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