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This paper proposes a novel architecture for the digital error correction logic block that is used in pipeline analog-to-digital converters. The new architecture is implemented with HA and OR_HA blocks instead of HA and FA in the conventional architecture. This architecture for digital error correction logic is simulated in 0.18µm CMOS process by using Cadence. The simulation results show that the proposed architecture improves speed and power consumption. Also this architecture occupies less area than the conventional digital error correction block.

this paper proposes a novel dual-stage resistor string digital to analog converter (DAC) architecture with a 12-bit resolution and less than 10ns settling time. The proposed architecture is constructed from the combination of two 6-bit RDACs in the first- and second stage with a reduced number of taps. In the proposed RDAC, the unity-gain buffer used for the isolation between the two stages is removed. The 12-bit resistor-string RDAC is designed using the 0.35-μm CMOS technology. The Mont Carlo simulations show a worst-case DNL = 0.2 LSB, INL = 0.3 LSB, and SNDR = 68.2 dB at 50MHz sampling frequency.

This paper introduces a novel structure for the comparator of analog – to -digital convertor with built- in threshold. Thus the resistor string can be eliminated. The proposed comparator is simulated using HSpice in TSMC 0.18μm CMOS process. The proposed structure reduces the power consumption and occupies less area compared to conventional comparators due to the elimination of the resistor string in analog-to-digital converters.

In this paper we introduce an Modified Clipped LMS (MCLMS) algorithm with a variable step size. In the MCLMS algorithm two parameters, the step size and the threshold control the convergence rate of the adaptive filter coefficients and also determine the final mean-square error. The computational complexity decreased dramatically by a large threshold. However, this selection results in a low convergence rate. Since the convergence time is inversely proportional to the step size, a large step size is often selected for fast convergence. But a large step size results in an increased final mean square error. Therefore in this paper we choose a large threshold and propose a variable step size for the MCLMS algorithm. The advantages of this proposed variable step size and a large threshold selection are that the computation complexity is low, final mean square error is low and that the convergence is fast.

In this paper, a new mathematical analysis is presented that shows the effects of cyclic prefix insertion on removing inter symbol interference (ISI) and inter carrier interference (ICI) in the discrete multi-tone (DMT) systems. In the DMT, for subcarriers modulating, the input data stream is divided into lower rate sub-streams. The ISI is eliminated almost entirely by adding a guard interval (GI) at the beginning of each DMT symbol. However, rather than using an empty GI, this interval is filled with the end part of the DMT symbol. Through this method ISI and ICI are avoided. In all the previous works, the authors have shown the above results by only matrix analysis, without any information insight in different steps, but in this work we analyze it by a new method via signal processing analysis for the first time. Therefore, one knows what will be happen for every symbol in each step.

In this paper, three new constellations (NQAM, CIRC1 and CIRC2) are introduced instead of the most popular quadrature amplitude modulation (QAM) in the discrete multi-tone (DMT) system. For the comparison purpose, they have been chosen so that they have the same average power. Also 8 standard channels (CSA#1 through CSA#8) with the AWGN noise have been employed. Based on the simulation results, the NQAM constellation has a better performance, in terms of the BER, with respect to popular QAM constellation without any increment in its complexity. The other two new constellations also have better performances with respect to the QAM in somewhere. Finally, for wirelined systems like the ADSL, the proposed NQAM is a good choice instead of the QAM.

In this paper an efficient quasi LMS/Newton adaptive algorithm is proposed for stereophonic acoustic echo cancellation. This method employs an efficient pseudo-diagonalization approach to the estimated joint-input correlation matrix with an emphasis on reducing its high cross-correlation components. We derive an estimate of the inverse joint-input correlation matrix by means of this pseudo-diagonalization method. Simulation results show the improvement in convergence performance of this algorithm compared to the nonlinear RLS and nonlinear NLMS adaptive algorithms.

In this paper, an ultra low power and low jitter 12bit CMOS digitally controlled oscillator (DCO) design is presented. Based on a ring oscillator implemented with low power Schmitt trigger based inverters. Simulation of the proposed DCO using 32nm CMOS Predictive Transistor Model (PTM) achieves controllable frequency range of 550MHz∼830MHz with a wide linearity and high resolution. Monte Carlo simulation demonstrates that the time-period jitter due to random power supply fluctuation is under 31ps and the power consumption is 0.5677mW at 750MHz with 1.2V power supply and 0.53-ps resolution. The proposed DCO has a good robustness to voltage and temperature variations and better linearity comparing to the conventional design.

In this paper, an all digital phase locked loop is proposed. The proposed ADPLL uses counter for locking the output signal. The proposed circuit has a simple structure in locking mechanism, and the power consumption of this design is very low. The proposed design is evaluated in TSMC 180nm and PTM 32nm. The power consumption of the proposed ADPLL for 180nm, at 403 MHz frequency is 2.5mW and in 32nm at 720MHz is 1.33 mW. Two novel blocks are introduced for glitch removing in ADPLL and any other digital circuit. The reduction of glitches also affects on low power design. The digitally controlled oscillator (DCO) used here is based on ring oscillators with fine and coarse bits applied on it. The DCO achieves very high resolution with 0.1 ps due to 16-bit control code.

Acoustic transmission for conferencing systems have progressed from the use of single channel to one that employs stereophonic channels. One of the most important challenges for such stereophonic system is the problem of stereophonic acoustic echo cancellation (SAEC) where a pair of echo cancellers are deployed to estimate the acoustic impulse responses of the receiving room. We propose, in this paper, a neural network based adaptive filtering approach for SAEC. The neural network is employed to decorrelate the input vectors for efficient filter updating, resulting in a high convergence rate of the adaptive filters for this multi-channel acoustic application. To further enhance the efficiency of the proposed algorithm, we then utilize the joint-input correlation matrix of the stereophonic signals so as to simplify the proposed neural network. Simulation results show the improvement in performance of the proposed adaptive SAEC approach over the state-of-the-art algorithms.

The use of partial-updating algorithm for reducing interchannel coherence in stereophonic acoustic echo cancellation has been proposed recently. In this work, we show that this algorithm suffers from the lack of robustness against source positions in the transmission room. To address this, we present an insight into this problem and propose a center clipping algorithm that improves the joint optimization between reducing interchannel coherence and increasing energies of selected taps for different source positions. Simulation results using both colored and speech inputs verify the robustness of the proposed algorithm against source positions.

Stereophonic acoustic echo cancellers (SAEC) have extended application in stereo teleconferencing systems. They have problems more severe than monophonic echo cancellers. The most important problem is the misalignment. This problem causes a severe divergence of filters in the event of abrupt changes of the transmission room acoustic paths. In this paper, we propose a novel wavelet thresholding approach for reducing the cross-correlation of the input vectors in the wavelet domain that leads to an improvement in the rate of misalignment reduction in adaptive filters employed in the stereophonic acoustic echo cancellation.

Stereophonic acoustic echo cancellers (SAEC) have problems more severe than monophonic echo cancellers. The most important problem is the misalignment, which is the mismatch between the impulse responses of adaptive filters and those of the acoustic paths of the receiving room (in the teleconferencing application, as an example) and therefore a severe divergence of filters in the event of abrupt changes of the transmission room acoustic paths. The main reason for such mismatching is the two-channel nature of the system and the high degree of cross-correlation of the input signals of those channels. In this paper, we try to reduce the correlation between the two channels in the transform domain. To achieve this, with inspiration from the lattice predictor structure, we present a novel approach for reduction of cross-correlation of the input vectors of two channels in the wavelet domain that lead to a reduction in the misalignment of adaptive filters for the SAEC.

In this paper we propose a new constellation that has better performance (in BER) with respect to custom QAM constellation. We compare the new one and two kinds of circular constellations with QAM. For comparison we chose them so that they have the same power average. Also we used FHT instead of FFT for all constellations and compared them with each other. Based on the results DMT, without cyclic prefix, with FHT has better performance with respect to FFT. Also the new constellation has a minor better performance in the channels which have maximum SNR=20 dB. Also it is very attractive that in the channel with SNR greater than 27 dB circular constellation (form II) has better performance with respect to form I especially when we use FHT.

Today, Internet access over a digital subscriber line (DSL) has been extended over the world. Recent technological advances have increased the potential capacity of the telephone operators' access networks, which were originally intended only to provide a PTSN. However, due to an imperfect balance between the twisted pair channels, different noises occur on the telephone lines. In this paper we apply different 16-QAM constellations (rectangular form and two non-rectangular form) to ADSL system with burst and normal noisy channel. The results of simulation show that rectangular constellations have better performance (especially BER) with respect to non- rectangular constellation in both normal and burst noisy channels.

In recent years, Internet access over a digital subscriber line (DSL) has experienced continuous growth. Recent technological advances have increased the potential capacity of the telephone operators' access networks, which were originally intended only to provide a plain old telephone service. However, due to an imperfect balance between the twisted pair channels, different noises occur on the telephone lines. In this paper we introduce new DMT system which is based on Hadamard transform instead of FFT to ADSL system with cyclic prefix and noisy channel. The results of simulation show that Hadamard have better performance (especially in BER) with respect to FFT for two out of three kind of simulations.

In recent years Internet access over a DSL has experienced continuous growth. Recent technological advances have increased the potential capacity of the telephone operators' access networks, which were originally intended only to provide a POTS. However, due to an imperfect balance between the twisted pair channels, different noises occur on the telephone lines. In this paper we apply Hadamard transform instead of FFT to ADSL system with 129 taps noisy channel. We did not use any cyclic prefix. The results of simulation show that Hadamard have better performance (especially BER) with respect to FFT. We introduce the new non-rectangular QAM that with FHT has better performance (in BER) with respect to rectangular QAM.

In recent years, Internet access over a digital subscriber line (DSL) has experienced continuous growth. Recent techno-logical advances have increased the potential capacity of the telephone operators' access networks, which were originally intended only to provide a plain old telephone service. However, due to an imperfect balance between the twisted pair channels, different noises occur on the telephone lines. In this paper we apply different 16-QAM constellations to ADSL system with burst noisy channel and the results of simulation show that non-rectangular constellations have better performance (especially BER) with respect to rectangular constellation.

This paper presents two new methods for designing digital filters using Fourier series expansion. The first method employs the impulse response of the filter, and the second method uses an FIR transform to obtain a digital filter from analog filter transfer function. Experimental results illustrate the accuracy of the methods.