1,695 research outputs found

    Design and multiplier-less implementation of a class of two-channel PR FIR filterbanks and wavelets with low system delay

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    In this paper, a new method for designing two-channel PR FIR filterbanks with low system delay is proposed. It is based on the generalization of the structure previously proposed by Phoong et al. Such structurally PR filterbanks are parameterized by two functions (β(z) and α(z)) that can be chosen as linear-phase FIR or allpass functions to construct FIR/IIR filterbanks with good frequency characteristics. The case of using identical β(z) and α(z) was considered by Phoong et al. with the delay parameter M chosen as 2N - 1. In this paper, the more general case of using different nonlinear-phase FIR functions for β(z) and α(z) is studied. As the linear-phase constraint is relaxed, the lengths of β(z) and α(z) are no longer restricted by the delay parameters of the filterbanks. Hence, higher stopband attenuation can still be achieved at low system delay. The design of the proposed low-delay filterbanks is formulated as a complex polynomial approximation problem, which can be solved by the Remez exchange algorithm or analytic formula with very low complexity. In addition, the orders and delay parameters can be estimated from the given filter specifications using a simple empirical formula. Therefore, low-delay two-channel PR filterbanks with flexible stopband attenuation and cutoff frequencies can be designed using existing filter design algorithms. The generalization of the present approach to the design of a class of wavelet bases associated with these low-delay filterbanks and its multiplier-less implementation using the sum of powers-of-two coefficients are also studied.published_or_final_versio

    Subband Adaptive Modeling of Digital Hearing Aids

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    In this thesis, the application of a subband adaptive model to characterize compression behaviour of five digital hearing aids is investigated. Using a signal-to-error ratio metric, modeling performance is determined by varying the number of analysis bands in the subband structure as well as consideration of three adaptive algorithms. The normalized least mean-squares (NLMS), the affine projection algorithm (APA), and the recursive least-squares (RLS) algorithms are employed using a range of parameters to determine the impact on modeling performance. Using the subband adaptive model to estimate the time-varying frequency response of each hearing aid allows the Perceptual Evaluation of Speech Quality (PESQ) mean-opinion score (MOS) to be computed. The PESQ MOS facilitates an estimation of a subjective assessment of speech quality using an objective score. Initial results suggest the PESQ MOS score is able to differentiate speech processed by hearing aids allowing them to be ranked accordingly. Further work is required to obtain subjective assessments of the processed speech signals and determine if possible correlations exist

    Multiplierless multirate FIR filter design and implementation

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    Ph.DDOCTOR OF PHILOSOPH

    A comparison of adaptive predictors in sub-band coding

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    Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1987.Bibliography: leaves 82-85.by Paul Ning.M.S

    Digital signal processing algorithms and structures for adaptive line enhancing

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    Imperial Users onl

    Linear Operation of Switch-Mode Outphasing Power Amplifiers

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    Radio transceivers are playing an increasingly important role in modern society. The ”connected” lifestyle has been enabled by modern wireless communications. The demand that has been placed on current wireless and cellular infrastructure requires increased spectral efficiency however this has come at the cost of power efficiency. This work investigates methods of improving wireless transceiver efficiency by enabling more efficient power amplifier architectures, specifically examining the role of switch-mode power amplifiers in macro cell scenarios. Our research focuses on the mechanisms within outphasing power amplifiers which prevent linear amplification. From the analysis it was clear that high power non-linear effects are correctable with currently available techniques however non-linear effects around the zero crossing point are not. As a result signal processing techniques for suppressing and avoiding non-linear operation in low power regions are explored. A novel method of digital pre-distortion is presented, and conventional techniques for linearisation are adapted for the particular needs of the outphasing power amplifier. More unconventional signal processing techniques are presented to aid linearisation of the outphasing power amplifier, both zero crossing and bandwidth expansion reduction methods are designed to avoid operation in nonlinear regions of the amplifiers. In combination with digital pre-distortion the techniques will improve linearisation efforts on outphasing systems with dynamic range and bandwidth constraints respectively. Our collaboration with NXP provided access to a digital outphasing power amplifier, enabling empirical analysis of non-linear behaviour and comparative analysis of behavioural modelling and linearisation efforts. The collaboration resulted in a bench mark for linear wideband operation of a digital outphasing power amplifier. The complimentary linearisation techniques, bandwidth expansion reduction and zero crossing reduction have been evaluated in both simulated and practical outphasing test benches. Initial results are promising and indicate that the benefits they provide are not limited to the outphasing amplifier architecture alone. Overall this thesis presents innovative analysis of the distortion mechanisms of the outphasing power amplifier, highlighting the sensitivity of the system to environmental effects. Practical and novel linearisation techniques are presented, with a focus on enabling wide band operation for modern communications standards

    DSP Based Transmitter I/Q Imbalance Calibration: Implementation and Performance Measurements

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    The recent interest in I/Q signal processing based transceivers has resulted in a new domain of research in flexible, low-power, and low-cost radio architectures. The main advantage of complex or I/Q up- and downconversion is that it does not produce any image signal and eliminates the need of expensive RF filters. This greatly simplifies the transceiver front-end and permits single-chip radio transceiver solutions. The analog quadrature modulators and demodulators are, however, sensitive to two kinds of implementation impairments: gain imbalance, and phase imbalance. These impairments originate due to the non-ideal behavior of the electronic components in the I- and Q- channels of the modulators/demodulators. As a result, they compromise the infinite image signal attenuation and adversely affect the performance of a wireless system. Furthermore, new higher order modulated waveforms and wideband signals are especially susceptible to these impairments and achieving sufficient image signal attenuation is a fundamental requirement for future wireless systems. Therefore, digital techniques which enhance the dynamic range of front-end with minimum amount of additional analog hardware are becoming more popular, being also motivated by the constantly increasing number crunching power of digital circuitry. In this thesis, some recently developed algorithms for I/Q imbalance estimation and compensation are studied on the transmitter side. The calibration algorithms use a baseband test signal combined with a feedback loop from I/Q modulator output back to transmitter digital parts to efficiently estimate the modulator I/Q mismatch. In the feedback loop, the RF signal is demodulated and compared with the original test signal to estimate the I/Q imbalance and the needed pre-distortion parameters. The actual digital transmit signal is then properly pre-distorted with the obtained I/Q imbalance knowledge, in order to cancel the effects of modulator I/Q imbalance at the data transmission phase. The performance of the compensation algorithms is first evaluated with computer simulations. A prototype system using laboratory instruments is also developed to illustrate the effects of I/Q imbalance in direct conversion and low-IF transmitters and is used to prove the usability of algorithms in real life front-ends. The results of computer simulations and laboratory measurements prove that the compensation algorithms yield a good calibration performance by suppressing the image signal interference close to or even below the noise floor. /Kir1
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