Geneviève Baudoin

EER architecture specifications for OFDM transmitter using a class E amplifier

This letter presents envelope elimination and restoration (EER) architecture specifications in the case of orthogonal frequency division multiplex C band transmission. A key point is the recombination of envelope and phase information by supply modulation of the power amplifier (PA). Imperfections, such as time mismatch and phase noise can reduce the performances of the transmission. Compression and conversion effects when supply modulating the PA are introduced in this letter with the simulation of a class E power amplifier. This amplifier was designed under HP-ADS using a nonlinear MESFET model. Results are reported in terms of error vector measurement and spectrum for two different numbers of sub-carriers 32 and 128 in 16-QAM and 64-QAM cases.

Segmental vocoder-going beyond the phonetic approach

The problem of very low bit rate segmental speech coding is addressed. The basic units are found automatically in the training database using temporal decomposition, vector quantization and multigrams. They are modelled by HMMs. The coding is based on recognition and synthesis. In single speaker tests, we obtained intelligible and naturally sounding speech at a mean rate of 211.2 b/s. In the end, future extensions of our scheme (diphone-like synthesis and speaker adaptation) as well as possible use of automatically derived units in recognition are discussed.

Adaptive polynomial pre-distortion for linearization of power amplifiers in wireless communications and WLAN

This paper presents a new adaptive pre-distortion algorithm for linearization of power amplifiers and its application to non-constant envelope modulations such as QAM or OFDM. The pre-distortion system is polynomial. The criterion is minimization of a mean square error criterion between the baseband-equivalent of the output of the real amplifier and the ideally amplified signal. The analytic expression of the gradient of the criterion has been calculated. A stochastic gradient algorithm is applied using this analytic expression. A special normalization of the coefficients of the polynomial predistortion has been proposed to improve the speed of convergence. The method has been tested on a class AB power amplifier with a baseband signal corresponding to filtered QPSK and OFDM modulations.

Influence of time and processing mismatches between phase and envelope signals in linearization systems using envelope elimination and restoration, application to Hiperlan2

This paper is a theoretical and experimental study of the influence of time and processing mismatches between the envelope and the phase of an OFDM in an envelope elimination and restoration (EER) linearization system. We give the theoretical expression of the power spectral density of the distorted signal in the case without intersymbol interference at the emitter. In the general case, we give the average power of the distortion as a function of the time mismatch. We also propose an approximation of the distorted signal as a delayed version of the original signal, that explains the observed rotations of the constellations at the receiver on the different OFDM carrier. Others effects of EER principle are also studied, such as limitation of envelope and phase bandwidth and non-linearity of the envelope restoration command.

Corpus based very low bit rate speech coding

This paper presents a new very low bit rate segmental speech coding approach applying speech recognition in the coder and corpus based speech synthesis in the decoder. The system uses a large corpus of speech signals that is searched to find a speech segment similar to the segment to be coded. The elementary acoustical units for recognition and synthesis are determined automatically by an unsupervised training method. This approach is an alternative to using phoneme-derived linguistic units. Very good results are obtained at an average bit rate of 400 bits/second for a corpus of about 1 hour of speech. We present an efficient method for finding the best synthesis unit taking into account the good concatenation of successive segments. The proposed organization of the speech segments in the corpus allows a very efficient search of the best unit.

Advances in Very Low Bit Rate Speech Coding Using Recognition and Synthesis Techniques

ALISP (Automatic Language Independent Speech Processing) units are an alternative concept to using phoneme-derived units in speech processing. This article describes advances in very low bit rate coding using ALISP units. Results of speaker-independent experiments are reported and speaker clustering using vector quantization is proposed. The improvements of speech re-synthesis using Harmonic Noise Model and dynamic selection of units are discussed.

PWM coding and filtering of an OFDM envelope signal in a C band EER transmitter architecture

This paper concerns the simulation of OFDM envelope path in an EER architecture. The envelope information is clipped to reduce the dynamic variation, PWM coded and low-pass filtered. The PWM coding is simulated with a modified Schmitt trigger with different hysteresis values. The reference voltage shape and frequency choice are discussed. The low-pass filter parameters depend on the frequency of the reference voltage signal and are summarized. The conclusion shows the tradeoff between spectral response and EVM results for choosing Fcomp (frequency of reference signal in the trigger) and h (hysteresis parameter). All simulations are done under HP-ADS software for 16-QAM modulation scheme.

System concepts dedicated to UWB transmitter

This paper introduces new system concepts for an UWB transmitter architecture dedicated to 3.1-5.1 GHz band. The latter is developed for an application targeting data rates up to 1 Mbps, associated with localization and positioning. First, a pulse shape analysis is done in order to select the best one for our spectrum boundary mask. Then, based on a windowed cosine pulse choice, a transmitter architecture is proposed and two critical blocks, the pulse generator and the antenna, are studied more in detail. Moreover, a link budget is done in order to correlate bandwidth, ranging and localization precision

Very Low Bit Rate Speech Coding: Comparison of Data-Driven Units with Syllable Segments

Very low bit-rate (VLBR) coding of speech offers the opportunity to test methods of automatic generation of sub-word units. This paper describes two approaches to VLBR coding: the first based on ALISP (Automatic Language Independent Speech Processing) techniques, the second based on syllable segments. Experimental results are reported on a database of one Czech professional speaker. The obtained rates for unit encoding were approximately 135 bps for the former approach and 62 bps for the latter. The quality was evaluated by measuring the logarithmic spectral distortion (computed on LPC-spectra), and in informal listening tests. Possible mutual profits of each technique to the other are discussed.

Time behavioral model for phase-domain ADPLL based frequency synthesizer

In this paper, we present a time behavioral model of a recently proposed phase-domain all-digital phase-locked loop (ADPLL) for RF applications. This model can be easily implemented, and results in a versatile and fast ADPLL simulator that enables to study many aspects of the PLL, e.g. transient responses, steady states, limit cycles, or to perform perturbation analysis. Moreover, we present a baseband analysis that allows computing the power spectral density from the instantaneous frequency obtained as the output of the behavioral model. Simulations illustrate the effectiveness of this new behavioral model.