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Author

Ali Beydoun

Other affiliations: Supélec, Télécom ParisTech, Sorbonne  ...read more
Bio: Ali Beydoun is an academic researcher from Lebanese University. The author has contributed to research in topics: Delta-sigma modulation & Digital filter. The author has an hindex of 7, co-authored 28 publications receiving 118 citations. Previous affiliations of Ali Beydoun include Supélec & Télécom ParisTech.

Papers
More filters
Journal ArticleDOI
TL;DR: The feasibility of a frequency-band-decomposition (FBD) ADC using continuous time bandpass sigma–delta modulators, even in the case of large analog mismatches is shown.
Abstract: Parallelism can be used to increase the bandwidths of ADC converters based on sigma---delta modulators. Each modulator converts a part of the input signal band and is followed by a digital filter. Unfortunately, solutions using bandpass sigma---delta modulators are very sensitive to the position of the modulators' central frequencies. This paper shows the feasibility of a frequency-band-decomposition (FBD) ADC using continuous time bandpass sigma---delta modulators, even in the case of large analog mismatches. The major benefit of such a solution, called extended-frequency-band-decomposition (EFBD) is its low sensitivity to analog parameters. For example, a relative error in the central frequencies of 4% can be accepted without significant degradation in the performance (other published FBD ADCs require a precision of the central frequencies better than 0.1%). This paper will focus on the performance which can be reached with this system, and the architecture of the digital part. The quantization of coefficients and operators will be addressed. It will be shown that a 14 bit resolution can be theoretically reached using 10 sixth-order bandpass modulators at a sampling frequency of 800 MHz which results in a bandwidth of 80 MHz centered around 200 MHz (the resolution depends on the effective quality factor of the filters of the analog modulators).

20 citations

Proceedings ArticleDOI
20 Oct 2009
TL;DR: This paper is an abstract of the whole project and presents the main results of the EFBD, allowing large mismatches in the analog modulators without performance degradation, at the price of a calibration of the digital stage.
Abstract: Frequency-Band-Decomposition (FBD) is a good candidate to increase the bandwidths of ADC converters based on Sigma-Delta modulators, especially in the context of software radio, where very large bands need to be converted. Each modulator processes a part of the input signal band and is followed by an adapted digital filter. A new solution, called Extended Frequency-Band-Decomposition (EFBD) has been proposed during the ANR VersaNUM project, allowing large mismatches in the analog modulators without performance degradation, at the price of a calibration of the digital stage. This paper is an abstract of the whole project and presents its main results.

13 citations

Proceedings Article
01 Sep 2006
TL;DR: This paper presents a new method for digitizing wideband signals based on the use of parallel analog delta sigma modulators, where each modulator converts a part of the input signal band and two solutions are proposed to reconstruct the signal.
Abstract: This paper presents a new method for digitizing wideband signals. It is based on the use of parallel analog delta sigma modulators, where each modulator converts a part of the input signal band. A major benefit of the architecture is that it widens the conversion band of the input signal and increases its dynamic range. Two solutions are proposed to reconstruct the signal: the first one uses bandpass filters without demodulation and the second demodulates the signal of each modulator, and then processes it in a lowpass filter. This paper focuses essentially on the digital part of the system and the overall performances are compared by using simulation results.

12 citations

Proceedings ArticleDOI
20 Oct 2009
TL;DR: A reconfigurable 4 channels TIΣΔ using the novel GMSCL (General Multi Stage Closed Loop) sigma-delta architecture and a new digital processing reducing considerably the hardware complexity is proposed.
Abstract: High performances wideband Analog to Digital Converter (ADC) remains a bottleneck to realize software and cognitive radio receivers. Time Interleaved Sigma-Delta (TIΣΔ) architecture is a good candidate to increase the bandwidth of the data converters with the lowest hardware complexity compared to other solutions using parallel sigma-delta modulators. This paper proposes a reconfigurable 4 channels TIΣΔ using the novel GMSCL (General Multi Stage Closed Loop) sigma-delta architecture and a new digital processing reducing considerably the hardware complexity. The sigma-delta modulators have been designed using switched-capacitor technique and implemented with STMicroelectronis 65 nm CMOS technology. Three different scenarios are possible : the first one for GSM standard clocked at 26 MHz and consumes 2.59 mW, the second one for UMTS/DVB-T standards clocked at 208 MHz and consumes 46 mW and the last one for WiFi/WiMax standards clocked at 208 MHz and consumes 92 mW. The total circuit die area is equal to 3 mm2. The digital filtering was validated and synthesized in a 1.2 V, 65 nm CMOS process using VHDL language. For a clock rate of 208 MHz, the evaluated die area is 0.115 mm2.

9 citations

Patent
28 Jul 2010
TL;DR: In this paper, the authors present an analog-to-digital converter comprising a time-interleaved multi-channel architecture, where digital filtering is applied in each channel, at least to estimate a converter offset error, and a compensation for the offset is applied on the basis of the estimated offset error.
Abstract: The present invention relates to signal processing in an analog-to-digital converter comprising a time-interleaved multi-channel architecture. According to the invention: digital filtering (H(z)) is applied in each channel, at least to estimate a converter offset error, and a compensation for the offset is applied on the basis of the estimated offset error. Advantageously, it is possible to benefit from the presence of a digital filter (H(z)) that is usually used to filter the quantization noise, such as a comb filter in converters that have sigma/delta modulators, in order to estimate the offset. The same filtering can then be applied in order to also estimate a gain disparity between the different channels.

9 citations


Cited by
More filters
Dissertation
04 Nov 2008
TL;DR: In this paper, the authors propose a solution to solve the problem of the problem: this paper ] of the "missing link" problem, i.i.p.II.
Abstract: II

655 citations

01 Jan 2016
TL;DR: This book helps people to understand why they end up in infectious downloads, rather than enjoying a good book with a cup of coffee in the afternoon, instead they are facing with some infectious virus inside their computer.
Abstract: Thank you for reading advances in network and acoustic echo cancellation. Maybe you have knowledge that, people have search hundreds times for their chosen books like this advances in network and acoustic echo cancellation, but end up in infectious downloads. Rather than enjoying a good book with a cup of coffee in the afternoon, instead they are facing with some infectious virus inside their computer.

122 citations

Patent
05 Jan 2011
TL;DR: In this article, a representative embodiment of an apparatus includes multiple quantization-noise-shaping continuous-time filters, each in a separate processing branch and having an adder that includes multiple inputs and an output.
Abstract: Provided are, among other things, systems, methods and techniques for converting a continuous-time, continuously variable signal into a sampled and quantized signal According to one representative embodiment, an apparatus includes multiple quantization-noise-shaping continuous-time filters, each in a separate processing branch and having an adder that includes multiple inputs and an output; an input signal is coupled to one of the inputs of the adder; the output of the adder is coupled to one of the inputs of the adder through a first filter; and the output of a sampling/quantization circuit in the same processing branch is coupled to one of the inputs of the adder through a second filter, with the second filter having a different transfer function than the first filter

97 citations

Journal ArticleDOI
TL;DR: In this article, the authors defined the power difference as the peak to average power ratio (PAPR) of the signal, which is defined as the difference between the maximum and average power of the input signal.
Abstract: Power efficiency is one of the most important parameters in designing communication systems, especially battery operated mobile terminals. In a typical transceiver, most of the power is dissipated in the power amplifier (PA) and consequently, it is very important to obtain the maximum efficiency from the PA. A PA operating in Class AB or B is at its maximum efficiency when it is driven by its maximum allowable input power [1]. In practice, the input signal of the PA usually has a varying envelope, and to avoid distortion the PA should not be driven to more than its maximum input saturating power. Unfortunately, this peak power of the input signal happens at very short periods, and most of the time the signal power is around its average power, which is much smaller than its peak power, meaning that, often, the PA works at much lower efficiencies than its maximum efficiency. The power difference is defined as the peak to average power ratio (PAPR) of the signal. For example, for a signal with 12 dB PAPR, a Class B PA would be driven with 12 dB power back-off from its peak input power, and at this power back-off, the efficiency of the PA will degrade from 78.5% to around 20% [1]. Unfortunately, by moving to high throughput modulation schemes, for example, quadrature amplitude modulations (QAMs) such as 16-QAM and 64-QAM mean that more envelope variation is needed to encode the information, and, consequently, lower efficiency is achieved.

59 citations

Patent
05 Jan 2011
TL;DR: In this article, the authors describe a system for converting a continuous-time, continuously variable signal into a sampled and quantized signal using a quantization-noise-shaping circuit, sampling/quantization circuit, and a digital bandpass filter.
Abstract: Provided are, among other things, systems, methods and techniques for converting a continuous-time, continuously variable signal into a sampled and quantized signal. According to one implementation, an apparatus includes multiple processing branches, each including: a continuous-time quantization-noise-shaping circuit, a sampling/quantization circuit, and a digital bandpass filter. A combining circuit then combines signals at the processing branch outputs into a final output signal. The continuous-time quantization-noise-shaping circuits include adjustable circuit components for changing their quantization-noise frequency-response minimum, and the digital bandpass filters include adjustable parameters for changing their frequency passbands.

41 citations