A new objective measurement system to predict speech intelligibility in binaural listening conditions is proposed for use with nonlinear hearing devices. Digital processing inside such devices often involves nonlinear operations such as clipping, compression, and noise reduction algorithms. Standard objective measures such as the Articulation Indeix (AI), the Speech Intelligibility Index (SII) and the Speech Transmission Index (STI) have been developed for monaural listening. Binaural extensions of these measures have been proposed in the literature, essentially consisting of a binaural pre-processing stage followed by monaural intelligibility prediction using the better ear or the binaurally enhanced signal. In this work, a three-stage extension of the binaural SII approach is proposed that deals with nonlinear acoustic input signals. The reference-based model operates as follows: (1) a stage to deal with nonlinear processing based on a signal-separation model to recover estimates of speech, noise and distortion signals at the output of hearing devices; (2) a binaural processing stage using the Equalization-Cancellation (EC) model; and (3) a stage for intelligibility prediction using the SII or the short-time Extended SII (ESII). Multiple versions of the model have been developed and tested for use with hearing devices. A software simulator is used to perform hearing-device processing under various binaural listening conditions. Details of the modeling procedure are discussed along with an experimental framework for collecting subjective intelligibility data. In the absence of hearing-device processing, the model successfully predicts speech intelligibility in all spatial configurations considered. Varying levels of success were obtained using two simple distortion modeling approaches with different distortion mechanisms. Future refinements to the model are proposed based on the results discussed in this work.

ITC/USA 2006 Conference Proceedings / The Forty-Second Annual International Telemetering Conference and Technical Exhibition / October 23-26, 2006 / Town and Country Resort & Convention Center, San Diego, California / Many satellite payloads require wide-band channels for transmission of large amounts of data to users on the ground. These channels typically have substantial distortions, including bandlimiting distortions and high power amplifier (HPA) nonlinearities that cause substantial degradation of bit error rate performance compared to additive white Gaussian noise (AWGN) scenarios. An adaptive equalization algorithm has been selected as the solution to improving bit error rate performance in the presence of these channel distortions. This paper describes the design and implementation of an adaptive baseband equalizer (ABBE) utilizing the latest FPGA technology. Implementation of the design was arrived at by first constructing a high fidelity channel simulation model, which incorporates worst-case signal impairments over the entire data link. All of the modem digital signal processing functions, including multirate carrier and symbol synchronization, are modeled, in addition to the adaptive complex baseband equalizer. Different feedback and feed-forward tap combinations are considered as part of the design optimization.

Ce travail concerne la linéarisation des amplificateurs de haute puissance en utilisant la pré-distorsion numérique. L’amplificateur de haute puissance est un composant non-linéaire. La pré-distorsion numérique adaptative en bande de base est un technique efficace pour linéariser ses non-linéarités et ses effets de mémoire. Les modèles de la pré-distorsion numérique de basse complexité sont étudiés dans cette thèse. Un algorithme est proposé pour déterminer une structure optimale de modèle uni-étage ou multi-étage en prenant compte du compromis entre la précision de modélisation et la complexité. La structure cascadée, qui est avantageuse en complexité comparé avec celle d'uni-étage, est étudiée avec des méthodes d'identifications différentes. En termes d'implémentations expérimentales, l'étude d'impact des choix de gain différents est approfondie dans cette thèse. Toutes les études ont été évaluées par un amplificateur de puissance Doherty / This dissertation contributes to the linearization techniques of high power amplifier using digital predistortion method. High power amplifier is one of the most nonlinear components in radio transmitters. Unfortunately, for most current types of power amplifiers, a good efficiency is obtained at the price of a poor linearity especially with modern communication waveforms. Baseband adaptive digital predistortion is a powerful technique to linearize the power amplifiers and allows to push the power amplifier operation point towards its high efficiency region. Linearization of power amplifiers using digital predistortion with low complexities is the focus of this dissertation. An algorithm is proposed to determine an optimal model structure of single-stage or multi-stage predistorter according to a trade-off between modeling accuracy and model complexity. Multi-stage cascaded digital predistortions are studied with different identification methods, which have advantages on complexity of model identification compared with single-stage structure. The linearization performances are validated by experimental implementations on test bench. In terms of experimental implementations, this dissertation studies the impact of different gain choices on linearized power amplifier. All studies are evaluated with a Doherty power amplifier

Surface acoustic wave (SAW) technology has been in use for well over one century. In the last few decades, due to its low cost and high performance, this technology has been widely adopted in modern wireless communication systems, to build filtering devices at radio frequency (RF). SAW filters and duplexers can be virtually found inside every mobile handset. SAW devices are traditionally recognized as passive devices with high linear signal processing behavior. However, recent deployments of third generation (3G) and fourth generation (4G) mobile networks require the handsets to handle an increasing number of frequency bands with more complex modulation /demodulation schemes and higher data rate for more subscribers. These requirements directly demand more stringent linearity specifications on the front end devices, including the SAW duplexers. In the past, SAW duplexer design was based on empirically obtained design rules to meet the linearity specifications. Lack of predictability and an understanding of the root cause of the nonlinearity have limited the potential applications of SAW duplexers. Therefore, research on the nonlinearity characterization and an accurate modeling of SAW nonlinearity for mobile device applications are very much needed. The Ph.D. work presented here primarily focuses on developing a general nonlinear model for SAW resonators/duplexers. Their nonlinear characteristics were investigated by measuring the harmonic and intermodulation distortions of resonators. A nonlinear Mason model is developed and the characterization results are integrated into SAW duplexer design flows to help to simulate the nonlinear effects accurately and improve the linearity performance of the products. iv In this dissertation, first, a novel nonlinear Mason equivalent circuit model including a third order nonlinear coefficient in the wave propagation is presented. Next, the nonlinear distortions of SAW resonators are analyzed by measuring large-signal harmonic and intermodulation spurious emission on resonators using a wafer probe station. The influence of the setups on the measurement reliability and reproducibility is discussed. Further, the nonlinear Mason model is validated by comparing its simulation results with harmonic and intermodulation measurements on SAW resonators and a WCDMA Band 5 duplexer. The Mason model developed and presented here is the first and only nonlinear physical model for SAW devices based on the equivalent circuit approach. By using this new model, good simulation measurement agreements are obtained on both harmonic and intermodulation distortions for SAW resonators and duplexers. These outcomes demonstrate the validity of the research on both the characterization and modeling of SAW devices. The result obtained confirms that the assumption of the representation of the 3 rd order nonlinearity in the propagation by a single coefficient is valid

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The objective of this research work is to investigate, design and measurement of a digital predistortion linearizer that is able to compensate the dynamic nonlinear distortion of a High Power Amplifier (PA). The effectiveness of the proposed baseband predistorter (PD) on the performance of a WLAN OFDM transmitter utilizing a nonlinear PA with memory effect is observed and discussed. For this purpose, a 10W Class-A/B power amplifier with a gain of 22 dB, operated over the 3.5 GHz frequency band was designed and implemented. The proposed baseband PD is independent of the operating RF frequency and can be used in multiband applications. Its operation is based on the Hammerstein system, taking into account PA memory effect compensation, and demonstrates a noticeable improvement compared to memoryless predistorters. Different types of modelling procedures and linearizers were introduced and investigated, in which accurate behavioural models of Radio Frequency (RF) PAs exhibiting linear and nonlinear memory effects were presented and considered, based on the Wiener approach employing a linear parametric estimation technique. Three new linear methods of parameter estimation were investigated, with the aim of reducing the complexity of the required filtering process in linear memory compensation. Moreover, an improved wiener model is represented to include the nonlinear memory effect in the system. The validity of the PA modelling approaches and predistortion techniques for compensation of nonlinearities of a PA were verified by several tests and measurements. The approaches presented, based on the Wiener system, have the capacity to deal with the existing trade-off between accuracy and convergence speed compared to more computationally complex behavioural modelling algorithms considering memory effects, such as those based on Volterra series and Neural Networks. In addition, nonlinear and linear crosstalks introduced by the power amplifier nonlinear behaviour and antennas mutual coupling due to the compact size of a MIMO OFDM transmitter have been investigated.

[EN] Common audio systems are designed with the intent of creating real and immersive scenarios that allow the user to experience a particular acoustic sensation that does not depend on the room he is perceiving the sound. However, acoustic devices and multichannel rendering systems working inside a room, can impair the global audio effect and thus the 3D spatial sound. In order to preserve the spatial sound characteristics of multichannel rendering techniques, adaptive filtering schemes are presented in this dissertation to compensate these electroacoustic effects and to achieve the immersive sensation of the desired acoustic system. Adaptive filtering offers a solution to the room equalization problem that is doubly interesting. First of all, it iteratively solves the room inversion problem, which can become computationally complex to obtain when direct methods are used. Secondly, the use of adaptive filters allows to follow the time-varying room conditions. In this regard, adaptive equalization (AE) filters try to cancel the echoes due to the room effects. In this work, we consider this problem and propose effective and robust linear schemes to solve this equalization problem by using adaptive filters. To do this, different adaptive filtering schemes are introduced in the AE context. These filtering schemes are based on three strategies previously introduced in the literature: the convex combination of filters, the biasing of the filter weights and the block-based filtering. More specifically, and motivated by the sparse nature of the acoustic impulse response and its corresponding optimal inverse filter, we introduce different adaptive equalization algorithms. In addition, since audio immersive systems usually require the use of multiple transducers, the multichannel adaptive equalization problem should be also taken into account when new single-channel approaches are presented, in the sense that they can be straightforwardly extended to the multichannel case. On the other hand, when dealing with audio devices, consideration must be given to the nonlinearities of the system in order to properly equalize the electroacoustic system. For that purpose, we propose a novel nonlinear filtered-x approach to compensate both room reverberation and nonlinear distortion with memory caused by the amplifier and loudspeaker devices. Finally, it is important to validate the algorithms proposed in a real-time implementation. Thus, some initial research results demonstrate that an adaptive equalizer can be used to compensate room distortions. / [ES] Los sistemas de audio actuales están diseñados con la idea de crear escenarios reales e inmersivos que permitan al usuario experimentar determinadas sensaciones acústicas que no dependan de la sala o situación donde se esté percibiendo el sonido. Sin embargo, los dispositivos acústicos y los sistemas multicanal funcionando dentro de salas, pueden perjudicar el efecto global sonoro y de esta forma, el sonido espacial 3D. Para poder preservar las características espaciales sonoras de los sistemas de reproducción multicanal, en esta tesis se presentan los esquemas de filtrado adaptativo para compensar dichos efectos electroacústicos y conseguir la sensación inmersiva del sistema sonoro deseado. El filtrado adaptativo ofrece una solución al problema de salas que es interesante por dos motivos. Por un lado, resuelve de forma iterativa el problema de inversión de salas, que puede llegar a ser computacionalmente costoso para los métodos de inversión directos existentes. Por otro lado, el uso de filtros adaptativos permite seguir las variaciones cambiantes de los efectos de la sala de escucha. A este respecto, los filtros de ecualización adaptativa (AE) intentan cancelar los ecos introducidos por la sala de escucha. En esta tesis se considera este problema y se proponen esquemas lineales efectivos y robustos para resolver el problema de ecualización mediante filtros adaptativos. Para conseguirlo, se introducen diferentes esquemas de filtrado adaptativo para AE. Estos esquemas de filtrado se basan en tres estrategias ya usadas en la literatura: la combinación convexa de filtros, el sesgado de los coeficientes del filtro y el filtrado basado en bloques. Más especificamente y motivado por la naturaleza dispersiva de las respuestas al impulso acústicas y de sus correspondientes filtros inversos óptimos, se presentan diversos algoritmos adaptativos de ecualización específicos. Además, ya que los sistemas de audio inmersivos requieren usar normalmente múltiples trasductores, se debe considerar también el problema de ecualización multicanal adaptativa cuando se diseñan nuevas estrategias de filtrado adaptativo para sistemas monocanal, ya que éstas deben ser fácilmente extrapolables al caso multicanal. Por otro lado, cuando se utilizan dispositivos acústicos, se debe considerar la existencia de no linearidades en el sistema elactroacústico, para poder ecualizarlo correctamente. Por este motivo, se propone un nuevo modelo no lineal de filtrado-x que compense a la vez la reverberación introducida por la sala y la distorsión no lineal con memoria provocada por el amplificador y el altavoz. Por último, es importante validar los algoritmos propuestos mediante implementaciones en tiempo real, para asegurarnos que pueden realizarse. Para ello, se presentan algunos resultados experimentales iniciales que muestran la idoneidad de la ecualización adaptativa en problemas de compensación de salas. / [CAT] Els sistemes d'àudio actuals es dissenyen amb l'objectiu de crear ambients reals i immersius que permeten a l'usuari experimentar una sensació acústica particular que no depèn de la sala on està percebent el so. No obstant això, els dispositius acústics i els sistemes de renderització multicanal treballant dins d'una sala poden arribar a modificar l'efecte global de l'àudio i per tant, l'efecte 3D del so a l'espai. Amb l'objectiu de conservar les característiques espacials del so obtingut amb tècniques de renderització multicanal, aquesta tesi doctoral presenta esquemes de filtrat adaptatiu per a compensar aquests efectes electroacústics i aconseguir una sensació immersiva del sistema acústic desitjat. El filtrat adaptatiu presenta una solució al problema d'equalització de sales que es interessant baix dos punts de vista. Per una banda, el filtrat adaptatiu resol de forma iterativa el problema inversió de sales, que pot arribar a ser molt complexe computacionalment quan s'utilitzen mètodes directes. Per altra banda, l'ús de filtres adaptatius permet fer un seguiment de les condicions canviants de la sala amb el temps. Més concretament, els filtres d'equalització adaptatius (EA) intenten cancel·lar els ecos produïts per la sala. A aquesta tesi, considerem aquest problema i proposem esquemes lineals efectius i robustos per a resoldre aquest problema d'equalització mitjançant filtres adaptatius. Per aconseguir-ho, diferent esquemes de filtrat adaptatiu es presenten dins del context del problema d'EA. Aquests esquemes de filtrat es basen en tres estratègies ja presentades a l'estat de l'art: la combinació convexa de filtres, el sesgat dels pesos del filtre i el filtrat basat en blocs. Més concretament, i motivat per la naturalesa dispersa de la resposta a l'impuls acústica i el corresponent filtre òptim invers, presentem diferents algorismes d'equalització adaptativa. A més a més, com que els sistemes d'àudio immersiu normalment requereixen l'ús de múltiples transductors, cal considerar també el problema d'equalització adaptativa multicanal quan es presenten noves solucions de canal simple, ja que aquestes s'han de poder estendre fàcilment al cas multicanal. Un altre aspecte a considerar quan es treballa amb dispositius d'àudio és el de les no linealitats del sistema a l'hora d'equalitzar correctament el sistema electroacústic. Amb aquest objectiu, a aquesta tesi es proposa una nova tècnica basada en filtrat-x no lineal, per a compensar tant la reverberació de la sala com la distorsió no lineal amb memòria introduïda per l'amplificador i els altaveus. Per últim, és important validar la implementació en temps real dels algorismes proposats. Amb aquest objectiu, alguns resultats inicials demostren la idoneïtat de l'equalització adaptativa en problemes de compensació de sales. / Fuster Criado, L. (2015). Linear and nonlinear room compensation of audio rendering systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/59459 / TESIS

The objective of this research work is to investigate, design and measurement of a digital predistortion linearizer that is able to compensate the dynamic nonlinear distortion of a High Power Amplifier (PA). The effectiveness of the proposed baseband predistorter (PD) on the performance of a WLAN OFDM transmitter utilizing a nonlinear PA with memory effect is observed and discussed. For this purpose, a 10W Class-A/B power amplifier with a gain of 22 dB, operated over the 3.5 GHz frequency band was designed and implemented. The proposed baseband PD is independent of the operating RF frequency and can be used in multiband applications. Its operation is based on the Hammerstein system, taking into account PA memory effect compensation, and demonstrates a noticeable improvement compared to memoryless predistorters. Different types of modelling procedures and linearizers were introduced and investigated, in which accurate behavioural models of Radio Frequency (RF) PAs exhibiting linear and nonlinear memory effects were presented and considered, based on the Wiener approach employing a linear parametric estimation technique. Three new linear methods of parameter estimation were investigated, with the aim of reducing the complexity of the required filtering process in linear memory compensation. Moreover, an improved wiener model is represented to include the nonlinear memory effect in the system. The validity of the PA modelling approaches and predistortion techniques for compensation of nonlinearities of a PA were verified by several tests and measurements. The approaches presented, based on the Wiener system, have the capacity to deal with the existing trade-off between accuracy and convergence speed compared to more computationally complex behavioural modelling algorithms considering memory effects, such as those based on Volterra series and Neural Networks. In addition, nonlinear and linear crosstalks introduced by the power amplifier nonlinear behaviour and antennas mutual coupling due to the compact size of a MIMO OFDM transmitter have been investigated.

<p>Most work on amplifier linearity has focused on the transconductance (gm) linearity, but there is increasing evidence that the voltage-dependence of the gate-source capacitance (Cgs) plays an important role in the linearity of emerging devices. This work addresses the capacitance contribution by incorporating the nonlinearities attributed to the voltage dependency of Cgs of a general FET on a circuit-level Cg amplifier model.</p> <p>An amplifier model including a voltage-dependent Cgs, and a voltage-dependent gm is studied using harmonic analysis and Volterra series. A closed form expression for the  third-order intercept point (IP3) of the amplifier, which depends on the nonlinear coefficients of Cgs, is obtained. A simple design rule, and a formula for the reduction of the IP3 due to the voltage-dependent Cgs are also presented. </p> <p>As application examples, the linearity of an amplifier based on a specific device is analyzed for two cases by extracting the nonlinear circuit parameters of the device. First for an analytic model of a bulk mosfet. Second for a one-dimensional, ballistic, coaxially gated Si nanowire. For low frequencies of design, the distortion introduced by gm is predominant, but for high frequencies it is obscured by the distortion coming from Cgs.</p> <p>We conclude that taking into account the voltage-dependence of Cgs is crucial when predicting the linearity behavior of a Cg amplifier, either designed for high-frequency operation, or based on a device operating near the quantum capacitance limit. </p>