Constant Envelope DCT- and FFT- based Multicarrier Systems

Discrete Cosine Transform (DCT)- and Fast Fourier Transform (FFT)- based Orthogonal Frequency Division Multiplexing (OFDM) systems with a variety of angle modulations are considered for data transmission. These modulations are used with the purpose of achieving Constant Envelope (CE) transmitted signals, for superior power efficiency with nonlinear High Power Amplifier (HPA), typically used at the transmitter in OFDM systems. Specifically, four angle modulations are considered: i) Phase Modulation (PM); ii) Frequency Modulation (FM); iii) Continuous Phase Modulation (CPM); and iv) Continuous Phase Chirp Modulation (CPCM). Descriptions of DCT- and FFT- based OFDM systems with M-ary Pulse Amplitude Modulation (MPAM) mapper, with these modulations, are given and expressions for transmitted signals are developed. The detection of these signals in Additive White Gaussian Noise (AWGN) and multipath fading channels is addressed. The receiver structure consists of arctangent demodulator followed by the optimum OFDM receiver for memoryless PM and FM modulations. However, for CPM and CPCM modulations that have inherent memory, arctangent demodulator followed by correction with oversampling technique is used prior to the optimum OFDM receiver. Closed-form expressions for Bit Error Rate (BER) have been derived and are function of: i) Signal-to-Noise Ratio, (Eb/N0); ii) Modulation parameters; iii) Number of amplitude levels of M-PAM mapper; and iv) parameters of multipath fading environment. It is shown that, in general, BER performance of CE-DCT-OFDM system is superior compared to that of conventional DCT-OFDM system, when the effect of HPA in the system is taken into account. Also, it is observed that CE-DCT-OFDM system outperforms CE-FFT-OFDM system by nearly 3 dB. The DCT- and FFT- OFDM systems with CPM and CPCM modulations are superior in BER performance compared to PM and FM modulations in these systems. The use of CPCM in OFDM systems can provide attractive trade off between bandwidth and BER performance. The performance of CE-DCT-OFDM and CE-FFT-OFDM systems over Rayleigh and Rician frequency non-selective slowly-varying fading channels are illustrated as a function of channel parameters and the penalty in SNR that must be paid as consequence of the fading is determined

On Efficient Signal Processing Algorithms for Signal Detection and PAPR Reduction in OFDM Systems

The driving force of the study is susceptibility of LS algorithm to noise. As LS algorithm is simple to implement, hence it’s performance improvement can contribute a lot to the wireless technology that are especially deals with high computation. Cascading of AdaBoost algorithm with LS greatly influences the OFDM system performance. Performance of Adaptive Boosting based symbol recovery was investigated on the performance of LS, MMSE, BLUE were also compared with the performance of AdaBoost algorithm and MMSE has been found the higher computational complexity. Furthermore, MMSE also requires apriori channel statistics and computational complexity O(5N3) of the MMSE increases exponentially as the number of carrier increases. For the Adaboost case the computational complexity calculation is little different.Therefore, in the training stage of the AdaBoost algorithm, the computational complexity is only O(nT M) Furthermore, as it is a classification algorithm so in the receiver side we will require a separate de-mapper (or decoder) to get the desired data bits, i.e., a. SAS aided DCT based PAPR reduction 1326 and b. SAS aided DCT based PAPR reduction. A successive addition subtraction preprocessed DCT based PAPR reduction technique was proposed. Here, the performance of proposed method was compared with other preexisting techniques like SLM and PTS and the performance of the proposed method was seen to outperform specially in low PAPR region. In the proposed PAPR reduction method, the receiver is aware of the transmitted signal processing, this enables a reverse operation at the receiver to extract the transmit data. Hence the requirement of sending extra information through extra subcarrier is eliminated. The proposed method is also seen to be spectrally efficient. In the case of PTS and SLM it is inevitable to send the side information to retrieve the transmit signal. Hence, these two methods are spectrally inefficient. Successive addition subtraction based PAPR reduction method was also applied to MIMO systems. The performance of the SAS based PAPR reduction method also showed better performance as compared to other technique. An extensive simulation of MIMO OFDM PAPR reduction was carried out by varying the number of subcarriers and number of transmitter antennas. A detailed computational complexity analysis was also carried out. BATE aided SDMA multi user detection. A detailed study of SDMA system was carried out with it’s mathematical analysis.Many linear and non linear detectors like ML, MMSE, PIC, SIC have been proposed in literature for multiuser detection of SDMA system. However, except MMSE every receivers other are computational extensive. So as to enhance the performance of the MMSE MUD a meta heuristic Bat algorithm was incorporated in cascade with MMSE

Exposing a waveform interface to the wireless channel for scalable video broadcast

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.Cataloged from PDF version of thesis.Includes bibliographical references (p. 157-167).Video broadcast and mobile video challenge the conventional wireless design. In broadcast and mobile scenarios the bit-rate supported by the channel differs across receivers and varies quickly over time. The conventional design however forces the source to pick a single bit-rate and degrades sharply when the channel cannot support it. This thesis presents SoftCast, a clean-slate design for wireless video where the source transmits one video stream that each receiver decodes to a video quality commensurate with its specific instantaneous channel quality. To do so, SoftCast ensures the samples of the digital video signal transmitted on the channel are linearly related to the pixels' luminance. Thus, when channel noise perturbs the transmitted signal samples, the perturbation naturally translates into approximation in the original video pixels. Hence, a receiver with a good channel (low noise) obtains a high fidelity video, and a receiver with a bad channel (high noise) obtains a low fidelity video. SoftCast's linear design in essence resembles the traditional analog approach to communication, which was abandoned in most major communication systems, as it does not enjoy the theoretical opimality of the digital separate design in point-topoint channels nor its effectiveness at compressing the source data. In this thesis, I show that in combination with decorrelating transforms common to modern digital video compression, the analog approach can achieve performance competitive with the prevalent digital design for a wide variety of practical point-to-point scenarios, and outperforms it in the broadcast and mobile scenarios. Since the conventional bit-pipe interface of the wireless physical layer (PHY) forces the separation of source and channel coding, to realize SoftCast, architectural changes to the wireless PHY are necessary. This thesis discusses the design of RawPHY, a reorganization of the PHY which exposes a waveform interface to the channel while shielding the designers of the higher layers from much of the perplexity of the wireless channel. I implement SoftCast and RawPHY using the GNURadio software and the USRP platform. Results from a 20-node testbed show that SoftCast improves the average video quality (i.e., PSNR) across diverse broadcast receivers in our testbed by up to 5.5 dB in comparison to conventional single- or multi-layer video. Even for a single receiver, it eliminates video glitches caused by mobility and increases robustness to packet loss by an order of magnitude.by Szymon Kazimierz Jakubczak.Ph.D

Scalable and rate adaptive wireless multimedia multicast

The methods that are described in this work enable highly efficient audio-visual streaming over wireless digital communication systems to an arbitrary number of receivers. In the focus of this thesis is thus point-to-multipoint transmission at constrained end-to-end delay. A fundamental difference as compared to point-to-point connections between exactly two communicating sending and receiving stations is in conveying information about successful or unsuccessful packet reception at the receiver side. The information to be transmitted is available at the sender, whereas the information about successful reception is only available to the receiver. Therefore, feedback about reception from the receiver to the sender is necessary. This information may be used for simple packet repetition in case of error, or adaptation of the bit rate of transmission to the momentary bit rate capacity of the channel, or both. This work focuses on the single transmission (including retransmissions) of data from one source to multiple destinations at the same time. A comparison with multi-receiver sequentially redundant transmission systems (simulcast MIMO) is made. With respect to feedback, this work considers time division multiple access systems, in which a single channel is used for data transmission and feedback. Therefore, the amount of time that can be spent for transmitting feedback is limited. An increase in time used for feedback transmissions from potentially many receivers results in a decrease in residual time which is usable for data transmission. This has direct impact on data throughput and hence, the quality of service. In the literature, an approach to reduce feedback overhead which is based on simultaneous feedback exists. In the scope of this work, simultaneous feedback implies equal carrier frequency, bandwidth and signal shape, in this case orthogonal frequency-division multiplex signals, during the event of the herein termed feedback aggregation in time. For this scheme, a constant amount of time is spent for feedback, independent of the number of receivers giving feedback about reception. Therefore, also data throughput remains independent of the number of receivers. This property of audio-visual digital transmission is taken for granted for statically configured, single purpose systems, such as terrestrial television. In the scope of this work are, however, multi-user and multi-purpose digital communication networks. Wireless LANs are a well-known example and are covered in detail herein. In suchlike systems, it is of great importance to remain independent of the number of receivers, as otherwise the service of ubiquitous digital connectivity is at the risk of being degraded. In this regard, the thesis at hand elaborates at what bit rates audio-visual transmission to multiple receivers may take place in conjunction with feedback aggregation. It is shown that the scheme achieves a multi-user throughput gain when used in conjunction with adaptivity of the bit rate to the channel. An assumption is the use of an ideal overlay packet erasure correcting code in this case. Furthermore, for delay constrained transmission, such as in so-called live television, throughput bit rates are examined. Applications have to be tolerant to a certain level of residual error in case of delay constrained transmission. Improvement of the rate adaptation algorithm is shown to increase throughput while residual error rates are decreased. Finally, with a consumer hardware prototype for digital live-TV re-distribution in the local wireless network, most of the mechanisms as described herein can be demonstrated.Die in vorliegender Arbeit aufgezeigten Methoden der paketbasierten drahtlosen digitalen Kommunikation ermöglichen es, Fernsehinhalte, aber auch audio-visuelle Datenströme im Allgemeinen, bei hoher Effizienz an beliebig große Gruppen von Empfängern zu verteilen. Im Fokus dieser Arbeit steht damit die Punkt- zu Mehrpunktübertragung bei begrenzter Ende-zu-Ende Verzögerung. Ein grundlegender Unterschied zur Punkt-zu-Punkt Verbindung zwischen genau zwei miteinander kommunizierenden Sender- und Empfängerstationen liegt in der Übermittlung der Information über erfolgreichen oder nicht erfolgreichen Paketempfang auf Seite der Empfänger. Da die zu übertragende Information am Sender vorliegt, die Information über den Erfolg der Übertragung jedoch ausschließlich beim jeweiligen Empfänger, muss eine Erfolgsmeldung auf dem Rückweg von Empfänger zu Sender erfolgen. Diese Information wird dann zum Beispiel zur einfachen Paketwiederholung im nicht erfolgreichen Fall genutzt, oder aber um die Übertragungsrate an die Kapazität des Kanals anzupassen, oder beides. Grundsätzlich beschäftigt sich diese Arbeit mit der einmaligen, gleichzeitigen Übertragung von Information (einschließlich Wiederholungen) an mehrere Empfänger, wobei ein Vergleich zu an mehrere Empfänger sequentiell redundant übertragenden Systemen (Simulcast MIMO) angestellt wird. In dieser Arbeit ist die Betrachtung bezüglich eines Rückkanals auf Zeitduplexsysteme beschränkt. In diesen Systemen wird der Kanal für Hin- und Rückweg zeitlich orthogonalisiert. Damit steht für die Übermittlung der Erfolgsmeldung eine beschränkte Zeitdauer zur Verfügung. Je mehr an Kanalzugriffszeit für die Erfolgsmeldungen der potentiell vielen Empfänger verbraucht wird, desto geringer wird die Restzeit, in der dann entsprechend weniger audio-visuelle Nutzdaten übertragbar sind, was sich direkt auf die Dienstqualität auswirkt. Ein in der Literatur weniger ausführlich betrachteter Ansatz ist die gleichzeitige Übertragung von Rückmeldungen mehrerer Teilnehmer auf gleicher Frequenz und bei identischer Bandbreite, sowie unter Nutzung gleichartiger Signale (hier: orthogonale Frequenzmultiplexsignalformung). Das Schema wird in dieser Arbeit daher als zeitliche Aggregation von Rückmeldungen, engl. feedback aggregation, bezeichnet. Dabei wird, unabhängig von der Anzahl der Empfänger, eine konstante Zeitdauer für Rückmeldungen genutzt, womit auch der Datendurchsatz durch zusätzliche Empfänger nicht notwendigerweise sinkt. Diese Eigenschaft ist aus statisch konfigurierten und für einen einzigen Zweck konzipierten Systemen, wie z. B. der terrestrischen Fernsehübertragung, bekannt. In dieser Arbeit werden im Gegensatz dazu jedoch am Beispiel von WLAN Mehrzweck- und Mehrbenutzersysteme betrachtet. Es handelt sich in derartigen Systemen zur digitalen Datenübertragung dabei um einen entscheidenden Vorteil, unabhängig von der Empfängeranzahl zu bleiben, da es sonst unweigerlich zu Einschränkungen in der Güte der angebotenen Dienstleistung der allgegenwärtigen digitalen Vernetzung kommen muss. Vorliegende Arbeit zeigt in diesem Zusammenhang auf, welche Datenraten unter Benutzung von feedback aggregation in der Verteilung an mehrere Empfänger und in verschiedenen Szenarien zu erreichen sind. Hierbei zeigt sich, dass das Schema im Zusammenspiel mit einer Adaption der Datenrate an den Übertragungskanal inhärent einen Datenratengewinn durch Mehrbenutzerempfang zu erzielen vermag, wenn ein überlagerter idealer Paketauslöschungsschutz-Code angenommen wird. Des weiteren wird bei der Übertragung mit zeitlich begrenzter Ausführungsdauer, z. B. dem sogenannten Live-Fernsehen, aufgezeigt, wie sich die erreichbare Datenrate reduziert und welche Restfehlertoleranz an die Übertragung gestellt werden muss. Hierbei wird ebenso aufgezeigt, wie sich durch Verbesserung der Ratenadaption erstere erhöhen und zweitere verringern lässt. An einem auf handelsüblichen Computer-Systemen realisiertem Prototypen zur Live-Fernsehübertragung können die hierin beschriebenen Mechanismen zu großen Teilen gezeigt werden