Tesi sul tema "Coded waveform"

Dans divers systèmes radar, un grand intérêt a été porté à la sélection d’une forme d’onde et à la conception d’une chaîne de traitement complète, de l’émetteur au récepteur, afin d’obtenir un profil distance haute résolution (HRRP, acronyme de High Range Resolution Profile en anglais). Au cours des dernières décennies, les concepteurs d’algorithmes de traitement du signal radar ont concentré leur attention sur différentes formes d’onde telles que les techniques de compression d’impulsion et les systèmes à bande synthétique (SF acronyme de stepped frequency, en anglais).D’une part, trois types de formes d’onde de compression d’impulsions large bande ont été proposés dans la littérature : la forme d’onde modulée linéairement en fréquence (Linear Frequency Modulation), celle à codes de phase (Phase Coded) et la forme d’onde modulé non linéairement en fréquence (Non Linear Frequency Modulation). Ces approches sont très populaires, mais elles requièrent une fréquence d’échantillonnage généralement élevée au niveau du récepteur, et par voie de conséquence un convertisseur analogique-numérique coûteux. De plus, les formes d’onde PC et NLFM peuvent être préférables dans certaines applications à haute résolution, car elles conduisent à de meilleurs PSLR et ISLR que ceux obtenus avec la forme d’onde LFM.D’autre part, lorsqu’il s’agit de schémas SF, une fréquence d’échantillonnage moins élevée peut être envisagée, ce qui permet d’utiliser un CAN meilleur marché.Ces deux approches peuvent être combinées pour tirer avantage des deux familles. Bien que la combinaison standard mène à l’exploitation d’un CAN bon marché, les performances en termes de PSLR et ISLR ne sont pas nécessairement adaptées. Comme le PSLR et l’ISLR ont une grande influence sur la probabilité de détection et la probabilité de fausse alarme, notre objectif est de trouver des solutions alternatives. Ainsi, notre contribution dans ce mémoire de thèse consiste à proposer deux nouvelles chaînes de traitement, de l’émetteur au récepteur :1) Dans la première approche, le spectre de la forme d’onde à large bande est décomposé en un nombre prédéterminé de portions. Puis, les versions temporelles de ces dernières sont successivement transmises. Le signal reçu est alors traité soit en utilisant un algorithme FD (pour Frequency domain en anglais) modifié, soit un algorithme de reconstruction de forme d’onde réalisé directement dans le domaine temporel (TWR pour time wave reconstruction). Dans cette thèse, les formes d’ondes PC et NLFM ont été sélectionnées. Une étude comparative est alors menée entre les différentes chaînes de traitement, de l’émetteur au récepteur, que l’on peut constituer. Nos simulations montrent que les performances obtenues à partir de l’algorithme TWR sont le plus souvent meilleures que celles de l’algorithme FD modifié. La contre-partie est une augmentation du coût calculatoire. De plus, que ce soit avec une forme d’onde PC ou NLFM, l’approche présentée fournit de meilleurs résultats en termes de PSLR et ISLR que les formes d’onde SF classiques.2) La seconde démarche proposée consiste à approximer une forme d’onde NLFM à large bande par une forme d’onde LFM par morceaux, puis de la combiner avec une approche de type SF. Cela donne lieu à une forme d’onde combinant SF et un train d’impulsions LFM ayant différentes durées et largeurs de bande. La sélection des paramètres de cette forme d’onde est faite en minimisant un critère multi-objectif, tenant compte du PSLR, de l’ISLR et de la résolution distance. Cette estimation est opérée par algorithmes génétiques. Selon les poids utilisés dans le critère multi-objectif et le nombre d’impulsions LFM pris en compte, les performances des les formes d’onde résultantes varient.Une annexe est en outre fournie qui présente des travaux complémentaires sur la comparaison de modèles à partir de la divergence de Jeffreys
In various radar systems, a great deal of interest has been paid to selecting a waveformand designing a whole processing chain from the transmitter to the receiver toobtain the high range resolution profile (HRRP). For the last decades, radar designershave focused their attentions on different waveforms such as the pulse compressionwaveforms and the stepped frequency (SF) waveform:On the one hand, three different types of wide-band pulse compression waveforms havebeen proposed: the linear frequency modulation (LFM) waveform, the phase coded(PC) waveform, and the non-linear frequency modulation (NLFM) waveform. They arevery popular but the sampling frequency at the receiver is usually large. This hence requiresan expensive analog-to-digital convertor (ADC). In addition, the PC and NLFMwaveforms may be preferable in some high range resolution applications since they leadto peak sidelobe ratio (PSLR) and integrated sidelobe ratio (ISLR) better than the onesobtained with the LFM waveform.On the other hand, when dealing with SF waveforms, a small sampling frequency canbe considered, making it possible to use a cheap ADC.Pulse compression and SF waveforms can be combined to take advantage of both. Althoughthe standard combination of PC or NLFM with SF leads to the exploitation ofa cheap ADC, the performance of the PC waveform or NLFM waveform in terms ofPSLR and ISLR cannot be attained. As the PSLR and the ISLR have a great influenceon the probability of detection and probability of false alarm, our purpose in the PhDdissertation is to present two new processing chains, from the transmitter to the receiver:1) In the first approach, the spectrum of a wide-band pulse compression pulse is splitinto a predetermined number of portions. Then, the time-domain transformedversions of these various portions are transmitted. At the receiver, the receivedechoes can be either processed with a modified FD algorithm or a novel timewaveformreconstruction (TWR) algorithm. A comparative study is carried outbetween the different processing chains, from the transmitter to the receiver, thatcan be designed. Our simulations show that the performance in terms of PSLRand ISLR obtained with the TWR algorithm is better than that of the modified FDalgorithm for a certain number of portions. This comes at the expense of an additionalcomputational cost. Moreover, whatever the pulse compression used, the approach we present outperforms the standard SF waveforms in terms of PSLRand ISLR.2) In the second approach, we suggest approximating the wide-band NLFM by apiecewise linear waveform and then using it in a SF framework. Thus, a variablechirp rate SF-LFM waveform is proposed where SF is combined with a train ofLFM pulses having different chirp rates with different durations and bandwidths.The parameters of the proposed waveform are derived from the wide-band NLFMwaveform. Then, their selection is done by considering a multi-objective optimization issue taking into account the PSLR, the ISLR and the range resolution.The latter is addressed by using a genetic algorithm. Depending on the weightsused in the multi-objective criterion and the number of LFM pulses that is considered, the performance of the resulting waveforms vary.An appendix is finally provided in which additional works are presented dealing withmodel comparison based on Jeffreys divergence

L'Internet des Objets (IoT), permettant l'interconnectivité de dispositifs physiques capables de collecter et d'échanger des données, connait depuis quelques années un immense intérêt. Son cercle d'influence touche de nombreux domaines très diverses comme par exemple la santé, l'agriculture, l'industrie ou bien encore les villes connectées.Cependant, le déploiement à grande échelle des équipements IoT peut présenter des difficultés, notamment pour des applications nécessitant des communications longue portée avec une faible consommation énergétique, tout en assurant des transmissions sans erreur dans des environnements très diverses. Les réseaux longue portée et à faible consommation énergétique (LPWANs) sont une solution qui proposent diverses technologies répondant aux contraintes des applications. L'une de ces technologies très populaires est la solution LoRa déployée dans les réseaux LoRa longue portée (LoRaWANs). Cette technologie basée notamment sur une modulation par étalement de spectre en utilisant un chirp (CSS), assure une transmission longue portée avec un faible débit et garantissant une bonne durée de vie des batteries des dispositifs IoT dans des environnements très contraignants. De plus, les codes correcteurs d'erreurs jouent un rôle primordial dans les communications modernes en détectant et corrigeant des erreurs de transmission en ajoutant de la redondance dans les messages envoyés. Pour les applications IoT, ces derniers peuvent éviter la retransmission de messages érronnés ou bien de diminuer la puissance d'émission, ce qui permet dans les deux cas de sauvegarder la batterie des équipements. Dans le contexte LoRaWAN, le choix de codes simples comme des codes de Hamming a été fait afin d'assurer un certain seuil de performance tout en ayant une complexité faible au niveau du décodage.Dans cette thèse, nous proposons tout d'abord une étude approfondie de la modulation LoRa et de nouvelles modulations basées sur LoRa proposées dans la littérature afin d'optimiser l'efficacité spectrale du signal. Ces nouvelles modulations proposent notamment l'ajout d'information dans la phase, l'emploi d'autres types de chirp ou bien l'utilisation de la composante en quadrature du signal. On s'intéresse aussi aux codes correcteurs d'erreurs pour ce type de modulation et pour des messages de petite taille dans divers environnements. On met notamment en lumière l'intérêt de schémas binaires codés itératifs utilisant des codes LDPC optimisés ou bien des schémas binaires codés à plusieurs étages (MLC) employant des codes polaires. Enfin, nous nous intéressons à des schémas codés non-binaires contraints par des ordres de modulations très élevés et des mots de code très courts en mettant en exergue le rapport performance/coût
The Internet of Things (IoT), enabling the interconnection of physical devices capable of collecting and exchanging data, has garnered immense interest in recent years. Its sphere of influence encompasses numerous diverse domains such as healthcare, agriculture, industry, and smart cities.However, the large-scale deployment of IoT devices can pose challenges, especially for applications requiring long-range communication with low power consumption, while ensuring error-free transmissions in highly diverse environments. Low-power wide area networks (LPWANs) offer a solution by providing various technologies that address application constraints. One such widely adopted technology is LoRa, deployed in LoRa wide area networks (LoRaWANs). This technology, based on chirp spread spectrum modulation (CSS), enables long-range transmission with low throughput, ensuring extended battery life for IoT devices in challenging environments.Furthermore, error-correcting codes play a pivotal role in modern communications by detecting and correcting transmission errors through redundancy in sent messages. For IoT applications, these codes can prevent the retransmission of erroneous messages or reduce transmission power, thereby conserving device battery life. In the context of LoRaWAN, the choice of simple codes such as Hamming codes has been made to ensure a certain performance threshold while maintaining low decoding complexity.In this thesis, we first propose a comprehensive study of LoRa modulation and new modulations based on LoRa proposed in the literature to optimize spectral efficiency. These new modulations include adding information in the phase, employing different types of chirps, or utilizing the quadrature component of the signal. We also investigate error-correcting codes for this type of modulation and for small message length in various environments. We particularly highlight the interest in iterative binary coded schemes using optimized LDPC codes or multi-level coded binary schemes employing polar codes. Finally, we delve into non-binary coded schemes constrained by very high modulation orders and very short codewords, emphasizing the performance-to-cost ratio

Thesis (MScEng)--Stellenbosch University, 2012.
ENGLISH ABSTRACT: In recent years Autonomous Underwater Vehicles (AUVs) have become interesting for harbor mapping and protection. AUVs require a SONAR sensor for observing their surroundings, thus enabling them to perform collision avoidance manoeuvres and scanning their operating environment for intruders or foreign objects, e.g. mines. To perform such actions the SONAR sensor is required to supply very fine range resolution for target imaging, as well as providing information about possible target velocity. Basic SONAR theory is discussed, as well as different approaches to signal design and processing techniques, for achieving the required resolution in range and target velocity. Two of the discussed approaches are selected for processing range and target velocity, respectively. Both approaches are simulated for their validity before being tested by using a custom-built platform. The platform is highly configurable and designed for capacity of testing a variety of SONAR signals and set ups. Furthermore, the platform is built by using off-the-shelf components to minimize development costs. The results of simulations and practical tests are presented. A high correlation between theory and practice is achieved. The knowledge and the platform presented form the stepping stone for further SONAR sensor developments.
AFRIKAANSE OPSOMMING: In die laaste jare het outonome onderwater voertuie (OOV) toenemend belangrik geword vir die kartografie en beskerming van hawens. OOV’s vereis SONAR sensore wat hulle in staat stel, om hulle omgewing waar te neem en sodoende botsing vermydings take te verrig en ook om hul werksomgewing noukeurig te skandeer om indringers of vreemde voorwerpe, bv. myne, op te spoor. Om sulke werk te verrig, word van die SONAR sensor vereis, om baie fyn afstand oplossings vir teiken te verskaf, insluitend die moontlike snelheid van die teiken. Basiese SONAR teorie word bespreek, en dan verskeie benaderings van sein ontwerp en verwerkings tegnieke. Twee van die bespreekte benaderings word gekies om afstand en teiken snelheid onderskeidelik te verwerk. Altwee benaderings word gesimuleer om hul geldigheid vas te stel, voor dat hulle getoets word op ’n pasmaat vervaardigde platform. Die platform is hoogs aanpasbaar en is ontwerp vir sy vermoë om ’n verskeidenheid SONAR seine en verwerkings te hanteer. Verder is die platform vervaardig met standard rakonderdele om ontwikkelingskoste so laag as moontlik te hou. Die uitslae van die simulerings en praktiese toetse word voorgestel. ’n Hoë mate aan korrelasie is bereik tussen teorie en praktyk. Die kennis en die platvorm, wat hier voorgestel word, vorm die eerste trappie vir toekomstige SONAR sensor ontwikkeling.

In this thesis study, an overview of MIMO radar is presented. The differences in radar cross section, channel and received signal models in different MIMO radar configurations are examined. The performance improvements that can be achieved by the use of waveform diversity in coherent MIMO radar and by the use of angular diversity in statistical MIMO radar are investigated. The optimal detector under Neyman-Pearson criterion for Coherent MIMO radar when the interfering signal is white Gaussian noise is developed. Detection performance of phased array radar, coherent MIMO radar and Statistical MIMO radar are compared through numerical simulations. A detector for MIMO radar that contains the space time codes explicitly is also examined.

Speech coding at bit rates near 4 kbps is expected to be widely deployed in applications such as visual telephony, mobile and personal communications. This research focuses on developing a speech coder based on the waveform interpolation (WI) scheme, with an attempt to deliver near toll-quality speech at rates around 4 kbps. A WI coder has been simulated in floating-point using the C programming language. The high performance of the WI model has been confirmed by subjective listening tests in which the unquantized coder outperforms the 32 kbps G.726 standard (ADPCM) 98% of the time under clean input speech conditions; the reconstructed speech is perceived to be essentially indistinguishable from the original. When fully quantized, the speech quality of the WI coder at 4.25 kbps has been judged to be equivalent to or better than that of G.729 (the ITU-T toll-quality 8 kbps standard) for 45% of the test sentences. Further refinements of the quantization techniques are warranted to bring the coder closer to the toll-quality benchmark. Yet, the existing implementation has produced good quality coded speech with a high degree of intelligibility and naturalness when compared to the conventional coding schemes operating in the neighbourhood of 4 kbps.

Approved for public release, distribution unlimited
The Joint Tactical Information Distribution System (JTIDS)) is a hybrid frequency-hopped, direct sequence spread spectrum system which used cyclic code-shift keying (CCFK) for M-ary symbol modulation and minimum shift-keying (MSK) for chip modulation. In addition JTIDS uses a (31, 15) Reed Solomon (RS) code for channel coding. In this thesis an alternative waveform consistent with the original JTIDS waveform is analyzed. The system to be considered uses a concatenated code consisting of a (31, k) Reed Solomon inner code and a 4/5 convolutional outer code. The coded symbols are transmitted on the in-phase (I) and quadrature (Q) components of the carrier using 32-ary orthogonal signaling with 32 chip basedband waveforms such as Walsh functions. Performance with both coherent and noncoherent detection is analyzed. For noncoherent detection only one five bit symbol is transmitted on the I and Q components of the carrier per symbol duration, so the data throughput for noncoherent detection 1/2 that of coherent detection. No diversity, consistent with JTIDS single-pulse structure, and a sequential diversity of two, consistent with JTIDS double-pulse structure, are both considered. For the double-pulse structure, performance is examined both for the case of linear soft diversity combining and also for soft diversity combining with perfectside information. Performance is examined for both AWGN only, as well as for AWGN and pulse-noise interference. Based on the results of this thesis, the proposed waveform is found to outperform the existing JTIDS/Link-16 waveform in all cases considered in this research. Indeed, the best performance for the atlernative waveform is obtained when an (31, 25) RD inner code is used. When only AWGN is present, the proposed waveform with no diversity has a gain of 2.6 dB and 2.5 dB as compared to the existing JTIDS/Link-16 wavefoorm for coherent and noncoherent demodulation, respectively, when Pb =10-5. Likewise in an AGWN only environment with a diversity of two, the proposed waveform outperforms the existing JTIDS/Link-16 waveform by 3.15 dB and 23 dB for coherent and noncoherent detection, respectively. When PNI is also present, the proposed waveform performs significantly better than the existing JTIDS waveform in all cases considered. Finally, the use of a concatenated code consisting of a (31, 25) RS inner code and a 4/5 convolutional outer code results in a 33% improvement in throughput as compared to the existing JTIDS/Link-16 waveform.

The goal of this thesis is to improve the quality of the Waveform Interpolation (WI) coded speech at 4.25 kbps. The quality improvement is focused on the efficient coding scheme of voiced speech segments, while keeping the basic coding format intact. In the WI paradigm voiced speech is modelled as a concatenation of the Slowly Evolving pitch-cycle Waveforms (SEW). Vector quantization is the optimal approach to encode the SEW magnitude at low bit rates, but its complexity imposes a formidable barrier.
Product code vector quantizers (PC-VQ) are a family of structured VQs that circumvent the complexity obstacle. The performance of product code VQs can be traded off against their storage and encoding complexity. This thesis introduces split/shape-gain VQ---a hybrid product code VQ, as an approach to quantize the SEW magnitude. The amplitude spectrum of the SEW is split into three non-overlapping subbands. The gains of the three subbands form the gain vector which are quantized using the conventional Generalized Lloyd Algorithm (GLA). Each shape vector obtained by normalizing each subband by its corresponding coded gain is quantized using a dimension conversion VQ along with a perceptually based bit allocation strategy and a perceptually weighted distortion measure. At the receiver, the discontinuity of the gain contour at the boundary of subbands introduces buzziness in the reconstructed speech. This problem is tackled by smoothing the gain versus frequency contour using a piecewise monotonic cubic interpolant. Simulation results indicate that the new method improves speech quality significantly.
The necessity of SEW phase information in the WI coder is also investigated in this thesis. Informal subjective test results demonstrate that transmission of SEW magnitude encoded by split/shape-gain VQ and inclusion of a fixed phase spectrum drawn from a voiced segment of a high-pitched male speaker obviates the need to send phase information.

Thesis (M.S. in Electronic Warfare Systems Engineering)--Naval Postgraduate School, September 2009.
Thesis Advisor(s): Robertson, Ralph C. "September 2009." Description based on title screen as viewed on 5 November 2009. Author(s) subject terms: Link-16/JTIDS, Reed-Solomon (RS) coding, Cyclic Code-Shift Keying (CCSK), Minimum-Shift Keying (MSK), convolutional codes, concatenated codes, perfect side information (PSI), Pulsed-Noise Interference (PNI), Additive White Gaussian Noise (AWGN), coherent detection, noncoherent detection. Includes bibliographical references (p. 79). Also available in print.

International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, Nevada
In this paper, a new coherent correlation-loop architecture for tracking direct-sequence spread-spectrum signals is proposed. In the proposed correlation loop model, the mean-square tracking error is minimized by varying the cross-correlation function between the received signal and the locally generated signal. The locally generated signal is produced by passing a replica of the transmitted signal through a linear time-invariant filter, which is termed the VCC filter. The issue of bandwidth of a correlation loop is addressed and a bandwidth definition for comparative purposes is introduced. The filter characteristics to minimize the tracking errors are determined using numerical optimization algorithms. This work demonstrates that the amplitude response of the VCC filter is a function of the input signal-to-noise ratio (SNR). In particular, the optimum filter does not replicate a differentiator at finite signal-to-noise ratio as is sometimes assumed. The optimal filter characteristics and the knowledge of the input SNR can be combined to produce a device that has very low probability of loosing lock.

Thesis (M.S. in Electrical Engineering)--Naval Postgraduate School, September 2009.
Thesis Advisor(s): Robertson, R. Clark. "September 2009." Description based on title screen as viewed on 6 November 2009. Author(s) subject terms: Link-16/JTIDS, (31, 15) Reed-Solomon (RS) coding, 32-ary Orthogonal signaling, Additive White Gaussian Noise (AWGN), Pulse-Noise Interference (PNI), Perfect Side Information (PSI). Includes bibliographical references (p. 49-51). Also available in print.

In this report metrics for jamming resistance and radar performance of waveform sets are described and developed, and different sets of waveforms are optimized, evaluated and compared. It is shown that without additional processing or PRI jitter, waveform sets can reach jamming resistance a few dB worse than what is provided by PRI jitter alone, and together with PRI jitter a few dB better. Waveforms with better jamming resistance tend to have worse range sidelobes and Doppler tolerance, but show less structure in their spectrograms, suggesting better LPI properties. The Doppler tolerance metric is new, as well as the comparative analysis of waveform sets on multiple metrics including jamming resistance.
Radar är fundamentalt i modern krigsföring. Med en radar kan man avfyra vapen från säkra avstånd och med precision mäta in mål. En radarstörare har som mål att förhindra en radar från att mäta in sitt mål. Då radarn fungerar genom att sända ut specifikt modulerade radiovågspulser och lyssna efter ekot från omgivningen kan störaren förhindra detta genom att antingen sända mycket starkt brus, eller genom att sända radiovågspulser med samma specifika modulation. Den senare metoden kallas för DRFM-störning, där förkortningen står för Digitalt RadioFrekvens-Minne, vilket antyder att störaren kan minnas radarns modulation och själv använda den. Om radarn använder en ny modulation (eng: waveform) för varje puls kan störaren inte använda modulationen den minns från förra pulsen utan måste vänta på att nästa puls träffar den innan den kan repetera pulsen, vilket begränsar dess störförmåga. Denna rapport tänker sig att radarn har en begränsad uppsättning av modulationer att byta mellan, och undersöker olika sådana uppsättningar och bedömer och jämför dem på olika mått av radarprestanda och störtålighet. Radioprestandamåtten inkluderar hur mycket förstärkning och hur fin upplösning man får av modulationen, hur väl modulationen kan hantera mycket snabba mål, och hur stora "sidolober" som uppstår runt starka mål. Sidolobsfenomenet är jämförbart med det optiska fenomenet där små men ljusstarka saker på natten kan se ut att ha en ljus halo eller ljusa utstrålningar runt sig. Störtålighetsmåtten kvantifierar hur distinkta de olika modulationerna i radarns uppsättning är, och på så vis hur väl radarn kan urskilja en modulation från de andra, tillsammans med hur liten sannolikheten är att störaren lyckas välja just den modulation vi kommer använda till nästa puls. Resultaten visar att metoden av modulationsbyten kan ge nästan lika stor störtålighet som en välkänd metod, PRI-jitter, ger själv och något högre i kombination med den metoden. Bättre störtålighet visas gå hand i hand med sämre mått på radarprestanda, men mindre strukturerade spektrogram vilket antyder att de kan vara svårare att upptäckas av radarspanare. Försämringen i måtten på radarprestanda innebär inte nödvändigtvis en lika stor försämring i faktisk radarprestanda, då sidoloberna tar an en brusartad karaktär vilket leder till praktiska fördelar gentemot de vanliga fixa sidoloberna.

ITC/USA 2009 Conference Proceedings / The Forty-Fifth Annual International Telemetering Conference and Technical Exhibition / October 26-29, 2009 / Riviera Hotel & Convention Center, Las Vegas, Nevada
Space-time Coding (STC) for Multiple-Input Multiple-Output (MIMO) wireless communication systems is an effective technique for providing robust wireless link performance in telemetry systems. This paper investigates the degradation in system performance when synchronization errors between the transmitter and receiver are present. Specifically, expressions that quantify the increase in symbol-error-rate as a function of symbol synchronization error are derived for a two-transmit and single receive antenna MISO system using binary frequency shift keying waveforms. These results are then extended to the MIMO case. The analytic results are verified with simulation results that show close agreement between the theoretical expressions and Monte Carlo simulation runs.

The past three decades has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for voice applications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. Then I proceed with descriptions of waveform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. At the end the system for computer-based stenographing is presented. Quality research and ways how to improve this system will be provided.

Thesis (M.S. in Electrical Engineering and M.S. in Electronic Warfare System Engineering)--Naval Postgraduate School, June 2008.
Thesis Advisor(s): Robertson, Clark. "June 2008." Description based on title screen as viewed on August 21, 2008. Includes bibliographical references (p. 39). Also available in print.

Divers radars sont développés pour des besoins d’aide à la conduite automobile de sécurité mais aussi de confort. Ils ont pour but de détecter la présence d’obstacles routiers afin d’éviter d’éventuelles collisions. La demande actuelle en termes de capteurs radars pour l’automobile connaît une croissance importante et les technologies employées doivent garantir de bonnes performances dans un environnement dégradé par les signaux interférents des autres utilisateurs. Dans cette thèse, nous nous intéressons au développement d’un système radar performant en tout lieu et en particulier dans un contexte multi-utilisateurs. A ce propos, nous proposons de nouvelles formes d’ondes qui se basent sur la combinaison des codes fréquentiels de Costas et d’autres techniques de compression d’impulsion en exploitant les signaux de Costas modifiés. La conception adoptée permet, grâce à la diversité introduite, de synthétiser un nombre important de formes d’ondes. Nous avons, ensuite, exploité deux approches d’estimation des paramètres des cibles. La première, plutôt classique, se base sur le traitement Doppler dans un train d’impulsions cohérent. La deuxième, récente dans le domaine automobile, se base sur la technique dite de « Compressed Sensing ». Une adaptation de ces algorithmes pour les signaux proposés a été discutée dans des environnements bruités et multi-cibles. L’ensemble de ces travaux contribue à explorer de nouvelles formes d’ondes, autres que celles utilisées dans les radars actuels et à proposer un traitement innovant en réception, adapté aux radars en général et à l’automobile en particulier
Several driver assistance radars are developed for security and comfort requirements. Their goal is among others to detect the presence of obstacles for collision avoidance. The current demand in terms of automotive radar sensors experience a significant growth and the technologies being employed must ensure good performances especially in an environment degraded by interfering signals of other users. In this thesis, we are interested in developing a radar system which is effective in all situations especially in a multi-user context. For this purpose, we propose novel radar waveforms based on the combination of frequency hopping Costas codes and other pulse compression techniques, using modified Costas signals. The design approach allows to synthesize a significant number of waveforms, thanks to the high diversity introduced. Afterwards, we have exploited two estimation of target parameters approaches. The first one, quite classic, is based on Doppler processing in a coherent pulse train. The second one, recent in the automotive field , is based on the Compressed sensing techniques. An adaptation of these algorithms to proposed signals is discussed in noisy and multi-target environments. All these works contribute in one hand to explore novel radar waveforms, complement to those currently used in automotive radars and in another hand to propose an innovative processing at the receiver level, suited to radar applications in general and automotive ones in particular

Aujourd’hui, les systèmes MSPSR (Multi-Static Primary Surveillance Radar) passifs se sont installés de manière durable dans le paysage de la surveillance aérienne [1]. L’intérêt que suscitent ces nouveaux systèmes provient du fait qu’en comparaison aux radars mono-statiques utilisés actuellement, les systèmes MSPSR reposent sur une distribution spatiale d’émetteurs et de récepteurs offrant des avantages en termes de fiabilité (redondance), de coûts (absence de joints tournants et émetteurs moins puissants) et de performances (diversité spatiale). Toutefois, le défaut majeur du MSPSR passif réside en l’absence de formes d’ondes dédiées due à l’exploitation d’émetteurs d’opportunités tels que les émetteurs de radio FM (Frequency Modulation) et/ou de DVB-T (Digital Video Broadcasting-Terrestrial) [2]. Afin de pallier à ce défaut, il est envisagé d’utiliser des émetteurs dédiés permettant l’emploi de formes d’ondes optimisées pour une application radar, on parle alors de MSPSR actif. Cette thèse se place dans ce cadre et a pour objectif d’étudier et de définir la ou les formes d’ondes ainsi que les traitements associés permettant d’atteindre de meilleurs performances : une meilleure flexibilité sur la disposition du système (positionnement des émetteurs libres), une continuité de service (non dépendance d’un système tiers) et de meilleurs performances radars (e.g. en terme de précision des mesures, détections, …). Dans ce but, cette thèse étudie : - Les critères de sélection des codes : comportement des fonctions d’ambiguïtés, PAPR (Peak to Average Power Ratio), efficacité spectrale, etc... ; - Les formes d’ondes utilisées en télécommunication (scrambling code, OFDM) afin d’identifier leur possible réemploi pour une application radar ; - L’utilisation d’algorithmes cycliques pour générer des familles de séquences adaptées à notre problème ; - Une approche basée sur une descente de gradient afin de générer des familles de codes de manière plus efficiente ; - Et l’évaluation des performances de ces différents algorithmes à travers l’établissement d’une borne supérieure sur le niveau maximum des lobes secondaires et à travers le dépouillement des données enregistrées suite à des campagnes d’essais
Nowadays, MSPSR (Multi-Static Primary Surveillance Radar) systems are sustainably settled in air surveillance program [1]. Compared to mono-static radar currently in use, an MSPSR system is based on a sparse network of transmitters (Tx) and receivers (Rx) interconnected to a Central Unit and offers advantages in terms of reliability, cost and performance.Two kinds of MSPSR systems exist: the Passive form and the Active one. While the Passive MSPSR uses transmitters of opportunity such as radio Frequency Modulation (FM) transmitters and/or Digital Video Broadcasting-Terrestrial (DVB-T) transmitters [2], the Active MSPSR uses dedicated transmitters, which emit a waveform that is controlled and designed for a radar application. Each receiver processes the signal coming from all transmitters and reflected on the targets; and the Central Unit restores the target location by intersecting “ellipsoids” from all (transmitter, receiver) pairs. Compared to passive MSPSR, the main advantages of the active MSPSR are the use of dedicated waveforms that allow reaching better performances (like a better association of the transmitters’ contributions at the receiver level); more flexibility in the deployment of transmitters and receivers station (in order to meet the requirements in localisation accuracy and in horizontal and altitude coverages); and the guarantee of having a service continuity. On this purpose, this thesis analyses the differents codes criteria such as the ambiguity function behaviour, the PAPR (Peak to Average Power Ratio), the spectrum efficiency, etc... . Then, in order to find dedicated waveforms for MSPSR systems, one solution is to find easily-constructed families of sequences. Thus building on the works carried out by the Telecommunication field for solving multi-user issues, this document investigates the application of spreading codes and OFDM signals in MSPSR concept. Besides, another solution is to directly generate a set of sequences. Based on cyclic algorithms in [3] we derive a new algorithm that allows to optimize sets of sequences. Similarly, using a gradient descent approach, we develop a more efficient algorithm than the cyclic one. Finally, in order to evaluate the performances of the different algorithms, this thesis generalizes the Levenshtein Bound, establishes new lower bounds on the PSLR (Peak Sidelobe Level Ratio) in mismatched filter case, and studies real data recorded during some trials

Aujourd’hui, les systèmes MSPSR (Multi-Static Primary Surveillance Radar) passifs se sont installés de manière durable dans le paysage de la surveillance aérienne [1]. L’intérêt que suscitent ces nouveaux systèmes provient du fait qu’en comparaison aux radars mono-statiques utilisés actuellement, les systèmes MSPSR reposent sur une distribution spatiale d’émetteurs et de récepteurs offrant des avantages en termes de fiabilité (redondance), de coûts (absence de joints tournants et émetteurs moins puissants) et de performances (diversité spatiale). Toutefois, le défaut majeur du MSPSR passif réside en l’absence de formes d’ondes dédiées due à l’exploitation d’émetteurs d’opportunités tels que les émetteurs de radio FM (Frequency Modulation) et/ou de DVB-T (Digital Video Broadcasting-Terrestrial) [2]. Afin de pallier à ce défaut, il est envisagé d’utiliser des émetteurs dédiés permettant l’emploi de formes d’ondes optimisées pour une application radar, on parle alors de MSPSR actif. Cette thèse se place dans ce cadre et a pour objectif d’étudier et de définir la ou les formes d’ondes ainsi que les traitements associés permettant d’atteindre de meilleurs performances : une meilleure flexibilité sur la disposition du système (positionnement des émetteurs libres), une continuité de service (non dépendance d’un système tiers) et de meilleurs performances radars (e.g. en terme de précision des mesures, détections, …). Dans ce but, cette thèse étudie : - Les critères de sélection des codes : comportement des fonctions d’ambiguïtés, PAPR (Peak to Average Power Ratio), efficacité spectrale, etc... ; - Les formes d’ondes utilisées en télécommunication (scrambling code, OFDM) afin d’identifier leur possible réemploi pour une application radar ; - L’utilisation d’algorithmes cycliques pour générer des familles de séquences adaptées à notre problème ; - Une approche basée sur une descente de gradient afin de générer des familles de codes de manière plus efficiente ; - Et l’évaluation des performances de ces différents algorithmes à travers l’établissement d’une borne supérieure sur le niveau maximum des lobes secondaires et à travers le dépouillement des données enregistrées suite à des campagnes d’essais
Nowadays, MSPSR (Multi-Static Primary Surveillance Radar) systems are sustainably settled in air surveillance program [1]. Compared to mono-static radar currently in use, an MSPSR system is based on a sparse network of transmitters (Tx) and receivers (Rx) interconnected to a Central Unit and offers advantages in terms of reliability, cost and performance.Two kinds of MSPSR systems exist: the Passive form and the Active one. While the Passive MSPSR uses transmitters of opportunity such as radio Frequency Modulation (FM) transmitters and/or Digital Video Broadcasting-Terrestrial (DVB-T) transmitters [2], the Active MSPSR uses dedicated transmitters, which emit a waveform that is controlled and designed for a radar application. Each receiver processes the signal coming from all transmitters and reflected on the targets; and the Central Unit restores the target location by intersecting “ellipsoids” from all (transmitter, receiver) pairs. Compared to passive MSPSR, the main advantages of the active MSPSR are the use of dedicated waveforms that allow reaching better performances (like a better association of the transmitters’ contributions at the receiver level); more flexibility in the deployment of transmitters and receivers station (in order to meet the requirements in localisation accuracy and in horizontal and altitude coverages); and the guarantee of having a service continuity. On this purpose, this thesis analyses the differents codes criteria such as the ambiguity function behaviour, the PAPR (Peak to Average Power Ratio), the spectrum efficiency, etc... . Then, in order to find dedicated waveforms for MSPSR systems, one solution is to find easily-constructed families of sequences. Thus building on the works carried out by the Telecommunication field for solving multi-user issues, this document investigates the application of spreading codes and OFDM signals in MSPSR concept. Besides, another solution is to directly generate a set of sequences. Based on cyclic algorithms in [3] we derive a new algorithm that allows to optimize sets of sequences. Similarly, using a gradient descent approach, we develop a more efficient algorithm than the cyclic one. Finally, in order to evaluate the performances of the different algorithms, this thesis generalizes the Levenshtein Bound, establishes new lower bounds on the PSLR (Peak Sidelobe Level Ratio) in mismatched filter case, and studies real data recorded during some trials

Thesis (Ph. D.)--Florida State University, 2003.
Advisor: Dr. Frank Gross, Florida State University, FAMU-FSU College of Engineering, Dept. of Electrical and Computer Engineering. Title and description from dissertation home page (viewed Oct. 3, 2003). Includes bibliographical references.

碩士
國立臺灣大學
電機工程學研究所
94
In medical ultrasound imaging, micro bubble contrast agents are used to improve the contrast between blood perfusion region and soft tissue. Micro bubble oscillation produces stronger nonlinear response than tissue. Pulse inversion (PI) imaging excites two phase inverted pulses and sums the echoes to cancel the linearly propagated signal and keep the nonlinear components. The imaging keeping fundamental part of residue by filtering is called PI fundamental imaging. Generally, PI fundamental imaging has better contrast to tissue ration than traditional fundamental imaging and second harmonic imaging. In this research, we use chirp excitation, which is one of coded waveforms as an attempt to improve CTR by increasing pulse length and maintain the axial resolution by received pulse compression. In this study, chirp excitation is applied to pulse inversion fundamental imaging and the effects of pulse, acoustic pressure, bubble radius, and compression filters on imaging compression are discussed. However, because the compression filter is designed assuming linear propagation, the range side lobe becomes significant in PI fundamental imaging because the signal is from the nonlinear response. The less strict filter constraint produces less high frequency noise and lower range side lobe in compression. By linear compression, the axial resolution recovery of PI fundamental image is 50 % less than fundamental image. Chirps with big initial frequency change such as frequency increasing or broad bandwidth chirp produce strong PI fundamental signal. Contrast agents having more bubbles with oscillation radius scatter stronger nonlinear signal. All of these increase image CTR. To sum up, using contrast agents with oscillation radius, Gaussian chirps, having high amplitude, decreasing frequency, and broad bandwidth, and less strict constraint filter can get image with best CTR and axial resolution. Our future work will focus on single bubble experiments and alternative pulse compression filter design.

The present thesis aims to make an in-depth study of Radar pulse compression, Neural Networks and Phase coded pulse compression codes. Pulse compression is a method which combines the high energy of a longer pulse width with the high resolution of a narrow pulse width. The major aspects that are considered for a pulse compression technique are signal to sidelobe ratio (SSR) performance, noise performance and Doppler shift performance. Matched filtering of biphase coded radar signals create unwanted sidelobes which may mask important information. The adaptive filtering techniques like Least Mean Square (LMS), Recursive Least Squares (RLS), and modified RLS algorithms are used for pulse radar detection and the results are compared. In this thesis, a novel approach for pulse compression using Recurrent Neural Network (RNN) is proposed. The 13-bit and 35-bit barker codes are used as signal codes to RNN and results are compared with Multilayer Perceptron (MLP) network. RNN yields better signal-to-sidelobe ratio (SSR), error convergence speed, noise performance, range resolution ability and Doppler shift performance than neural network (NN) and some traditional algorithms like auto correlation function(ACF) algorithm. But the SSR obtained from RNN is less for most of the applications. Hence a Radial Basis Function (RBF) neural network is implemented which yields better convergence speed, higher SSRs in adverse situations of noise and better robustness in Doppler shift tolerance than MLP and ACF algorithm. There is a scope of further improvement in performance in terms of SSR, error convergence speed, and Doppler shift. A novel approach using Recurrent RBF is proposed for pulse radar detection, and the results are compared with RBF, MLP and ACF. Biphase codes, namely barker codes are used as inputs to all these neural networks. The disadvantages of biphase codes include high sidelobes and poor Doppler tolerance. The Golay complementary codes have zero sidelobes but they are poor Doppler tolerant as that of biphase codes. The polyphase codes have low sidelobes and are more Doppler tolerant than biphase codes. The polyphase codes namely Frank, P1, P2, P3, P4 codes are described in detail and autocorrelation outputs, phase values and their Doppler properties are discussed and compared. The sidelobe reduction techniques such as single Two Sample Sliding Window Adder (TSSWA) and double TSSWA after the autocorrelator output are discussed and their performances for P4 code are presented and compared. Weighting techniques can also be applied to substantially reduce the range time sidelobes. The weighting functions such as Kaiser-Bessel amplitude weighting function and classical amplitude weighting functions (i.e. Hamming window) are described and are applied to the receiver waveform of 100 element P4 code and the autocorrelation outputs, Peak Sidelobe Level (PSL), Integrated Sidelobe Level (ISL) values are compared with that of rectangular window. The effects of weighting on the Doppler performance of the P4 code are presented and compared.

碩士
國立成功大學
電腦與通信工程研究所
103
In recent years, optical code-division multiple-access (OCDMA) systems have been proposed for multiple accesses to utilize the vast bandwidth available in optical fiber. Optical CDMA systems are believed to provide asynchronous access for each user in the system, which is especially suitable for usage in local area network (LAN). We propose an optical code-division multiple-access (SPE-OCDMA) codecs scheme for proof-of-concept demonstration. The spectrally phase- codecs are constructed with Liquid-Crystal Spatial-Light Modulator (LC-SLM), Mach-Zehnder Interferometer (MZI), and differential balanced photo-detectors. Spectrally phase encoding (SPE) scheme of optical CDMA has been introduced to eliminate the MAI effect by using the orthogonal coding, such as Walsh-Hadamard codes. However, the transmission processes will produce phase shift between transceiver codecs. The phase shift will destroy orthogonality of the signal codes and cause detection error in the receiver. There were articles dealt with solutions using optical phase- locked loop (OPLL), but it’s not easy to implement. We consider the effect of phase shift in the SPC-OCDMA system and good result is obtained with little system degradation.

Direct-sequence code-division multiple-access (DS-CDMA) technology is a popular communication scheme adopted for the third generation (3G) wireless communication systems and beyond. Computationally efficient linear multiuser detection techniques can significantly improve the capacity of DS-CDMA systems by taking into account not only the signature of the user-of-interest but also those of the interfering users. However, as the spread spectrum DS-CDMA signals are usually subject to frequency-selective fading, a mismatch between the presumed and the actual user signatures is a typical challenge at the receiver side. As such signature mismatch may have a substantially detrimental effect on the performance of multiuser receivers, it is required to use an accurate signature estimation technique prior to the subsequent multiuser detection procedure. Among various signature estimation approaches, the bandwidth efficient blind algorithms that do not require transmission of any training symbols are of significant interest. This fact have motivated the topic of this thesis where advanced blind signature estimation and multiuser detection techniques for DS-CDMA systems are developed and studied. The first part of the thesis is devoted to blind signature estimation. We first develop a novel subspace-based signature estimation technique for binary phase shift keying (BPSK) transmitted signals in a white noise scenario. Unlike the earlier conventional signature estimation techniques, our method exploits the received signal jointly with its complex conjugate to substantially improve the estimation performance. We then analyze the performance of two well-known subspace-based estimation techniques developed for the case of unknown correlated noise. Finally, we propose a new signature estimation technique for this case that, unlike the latter two estimation algorithms, can be applied to an arbitrary signal constellation and enjoys much simpler implementation. The second part of the thesis is devoted to blind linear multiuser receivers. In this part of our work, we first develop a robust blind minimum mean-square error (MMSE) receiver that explicitly accounts for both the errors in the estimated signature and the sample data covariance matrix. Next, we analyze the signal-to-interference-plus-noise ratio (SINR) performance of the popular blind minimum output energy (MOE) receiver in large random DS-CDMA communication systems. The asymptotic properties of the Capon receiver and the blind Capon channel estimate are also studied in detail for both uplink and downlink scenarios. Our study of existing blind signature estimation and multiuser detection algorithms reveal some important properties of these techniques that have been missing in the literature. Moreover, our novel algorithms meet the requirements of the evolving communication network standards for highly reliable yet simple detection strategies. As such, we expect that our results will facilitate the application of blind multiuser detection to the future generations of DS-CDMA communication systems.

by Ge Gao.
Thesis (M.Phil.)--Chinese University of Hong Kong, 1999.
Includes bibliographical references (leaves 101-107).
Text in English; abstracts in English and Chinese.
by Ge Gao.
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Attributes of speech coders --- p.1
Chapter 1.1.1 --- Bit rate --- p.2
Chapter 1.1.2 --- Speech quality --- p.3
Chapter 1.1.3 --- Complexity --- p.3
Chapter 1.1.4 --- Delay --- p.4
Chapter 1.1.5 --- Channel-error sensitivity --- p.4
Chapter 1.2 --- Development of speech coding techniques --- p.5
Chapter 1.3 --- Motivations and objectives --- p.7
Chapter 2 --- Waveform interpolation speech model --- p.9
Chapter 2.1 --- Overview of speech production model --- p.9
Chapter 2.2 --- Linear prediction(LP) --- p.11
Chapter 2.3 --- Linear-prediction based analysis-by-synthesis coding(LPAS) --- p.14
Chapter 2.4 --- Sinusoidal model --- p.15
Chapter 2.5 --- Mixed Excitation Linear Prediction(MELP) model --- p.16
Chapter 2.6 --- Waveform interpolation model --- p.16
Chapter 2.6.1 --- Principles of waveform interpolation model --- p.18
Chapter 2.6.2 --- Outline of a WI coding system --- p.25
Chapter 3 --- Pitch detection --- p.31
Chapter 3.1 --- Overview of existing pitch detection methods --- p.31
Chapter 3.2 --- Robust Algorithm for Pitch Tracking(RAPT) --- p.33
Chapter 3.3 --- Modifications of RAPT --- p.37
Chapter 4 --- Development of a 1.7kbps speech coder --- p.44
Chapter 4.1 --- Architecture of the coder --- p.44
Chapter 4.2 --- Encoding of unvoiced speech --- p.46
Chapter 4.3 --- Encoding of voiced speech --- p.46
Chapter 4.3.1 --- Generation of PCW --- p.48
Chapter 4.3.2 --- Variable Dimensional Vector Quantization(VDVQ) --- p.53
Chapter 4.3.3 --- Sparse frequency representation(SFR) of speech --- p.56
Chapter 4.3.4 --- Sample selective linear prediction (SSLP) --- p.58
Chapter 4.4 --- Practical implementation issues --- p.60
Chapter 5 --- Development of a 2.0kbps speech coder --- p.67
Chapter 5.1 --- Features of the coder --- p.67
Chapter 5.2 --- Postfiltering --- p.75
Chapter 5.3 --- Voice activity detection(VAD) --- p.76
Chapter 5.4 --- Performance evaluation --- p.79
Chapter 6 --- Conclusion --- p.85
Chapter A --- Subroutine for pitch detection algorithm --- p.88
Chapter B --- Subroutines for Pitch Cycle Waveform(PCW) generation --- p.96
Chapter B.1 --- The main subroutine --- p.96
Chapter B.2 --- Subroutine for peak picking algorithm --- p.98
Chapter B.3 --- Subroutine for encoding the residue (using VDVQ) --- p.99
Chapter B.4 --- Subroutine for synthesizing PCW from its residue --- p.100
Bibliography --- p.101