Adaptive, differential pulse-code modulation for speech processing

The objective of the research reported here is the design of efficient speech coders that can easily be implemented in integrated circuit hardware. Companding techniques like those introduced by M. R. Winkler, J. A. Greefkes, F. DeJager, A. Tomozawa and H. Kaneko were explored along with a large body of theory concerning the application of linear prediction to speech coding. The best features of the speech signal to be measured and coded are the overall amplitude, the resonant frequencies and dampings of the vocal cavity and the fundamental frequency of the vocal cord oscillations. Adaptive quantization was used to track variations in overall amplitude, and adaptive prediction was used to track the frequencies and dampings of the cavity resonances. No attempt was made to exploit redundancies related to the vocal cord oscillations, however. An adaptive differential pulse code modulator (i.e., an ADPCM coder) with a fixed integrator was simulated first. Later a hardware model was constructed, signal to noise measurements were taken and subjective tests conducted. When operating at 4 bits per sample, speech of a quality nearly equal to that of 7 bit log PCM was regenerated by the ADPCM encoder. At 3 bits per sample speech quality was nearly equal to 6 bit log PCM. Further improvements were achieved with the application of adaptive predictors in place of the integrator. The predictor coefficients form a vector which is adapted in a direction away from the gradient with respect to the error power. By applying this technique to the quantized signals occurring in the coder, the coefficients are derived from the quantized error signal; hence, there is no need to transmit them

An investigation into the use of orthogonal frequency division multiplexing in packet radio

Bibliography: leaves 56-58.The concept of Orthogonal Frequency Division Multiplexing has been around since the 1960s. It has resurfaced over the last decade as being the modulation scheme of choice in some newer technologies like Digital Video Broadcasting (DVB) and Asynchronous Digital Subscriber lines (ADSL). Amateur packet radio started in 1978 and has attracted thousands of enthusiasts from around the world. The interest in packet radio has waned over the years due advances in the data transmission capabilities oflandline systems and also more widespread access to the Internet. The purpose of this thesis was to develop a simple software simulation model to determine whether or not OFDM could be used to increase the data rates currently available in packet radio systems. The thesis starts out with an introduction to packet radio and OFDM in Chapter 1. A slightly more detailed discussion on OFDM is given in Chapter 2 in order to develop a basic specification for the proposed OFDM model. Chapters 2,3 and 4 discuss the development of the Transmitter model, the Receiver model and the Channel model respectively using the Simulink software package. Chapter 6 discusses the problem of Peak-to-Average Power Ratios (PAPR) in OFDM and explores the use of A-law companding to reduce this problem. In Chapter 7, the developed models are simulated and their performance compared to theoretical expectations. The full system is also simulated in this chapter in order to ascertain the possible data rate through the modelled packet radio channel. Conclusions regarding the application of OFDM to packet radio are presented in Chapter 8

Novel Bandpass Filter Design based on Synchronous Filtering

The design of high-performance low-noise bandpass filtering systems has been studied from several aspects: (1) applying the synchronous filtering idea to the development of externally linear, time-invariant filters which can be internally nonlinear and/or time-varying, (2) seeking solutions to improve the noise performance of these filters, from parameter configuration to architecture design, and (3) implementing the systems of interest as transistor level circuits and verifying their function.Particularly, the state space representations for a biquad AM mode synchronous bandpass filter and a biquad FM mode synchronous complex filter have been proposed and realized with ideal Gm-C networks and log-domain circuits. Both systems utilize the modulator-core filter-modulator architecture to synchronize the internal signal processing. The core filter in an AM mode synchronous filter has constant center frequency and time-variant bandwidth, and the terminal modulators perform amplitude modulation to maintain the system’s external linearity and input/output characteristics. An FM mode synchronous filter typically has time-invariant bandwidth and performs frequency modulation before and after the signal filtering. Depending on whether the center frequency and terminal modulating frequency vary with time, there are static and dynamic types of FM mode synchronous filters. They both have the advantage of being able to filter the high frequency input signals in a low frequency range, which greatly alleviates the design and integration challenge due to the high frequency limitation of active components. Moreover, some dynamic filters effectively suppress the injected single-tone noise and generate an output with much higher SNR in comparison to the output from a static filter that implements the same transfer function.As a variation of an AM mode synchronous bandpass filter, the system derived by removing its back end modulator has been verified to have impressive noise reduction capability when processing noisy AM signals. Furthermore, it inspired the development of a feedback filtering system, the effective bandwidth of which could be tuned by scaling the feedback signal that time varies the core filter’s instantaneous bandwidth. It further provides an innovative approach to the design of a high-Q filter with superior immunity to internal noise, using a filter with very low Q factor. Finally, a design that combines the feedback architecture and the biquad FM mode synchronous complex filter is proposed and implemented as a log-domain filtering circuit. Appealing features of this system include wide dynamic range, flexible bandwidth and center frequency tunability. Since there is a low requirement for the high-frequency performance of active components, these filters make a good fit for monolithic integration, and greatly improved immunity to in-filter noise in comparison to that of an open loop complex filter with similar external filtering capability

Design techniques for sigma-delta modulators in communications applications

Specialised design techniques for sigma-delta modulators are described in this thesis with all of the examples coming from modern communications systems. The noise shaping and the signal transfer functions can be optimised using a weighted least squares approach. Numerical problems arising in the optimisation as a result of high oversampling rates are overcome through the use a simple transformation. The application to digitising audio is discussed, with the conclusion that Butterworth response noise shaping is preferable to inverse Chebyshev noise shaping for audio applications. An example of optimising the signal transfer function to provide immunity to instability brought about by large out-of-band signals is also presented. The use of redundant arithmetic in the implementation of very high speed sigma-delta modulators is introduced, together with a DAC / filter combination suitable for reconstructing an analogue signal from the redundant arithmetic SDM. An improved topology for a speech compander is described which offers a number of significant advantages over existing published methods. This uses no external components for ac coupling or setting the response time-constant, yet is robust and insensitive to parasitic components and process variations. This has been integrated on a CMOS IC process and the results are compared with the high level simulations. A simulation method which allows the verification of switched-capacitor schematics with several orders of magnitude speed improvements over commercially available simulation tools is discussed. The method assumes ideal components, with internally controllable switches and reduces the schematic netlist to the few key equations that an experienced designer would derive manually. This process is fully automated and consequently is useful for providing confidence in implementations of complex SC systems

The low bit-rate coding of speech signals

Imperial Users onl

Distributed Digital Radios for Land Mobile Radio Applications

The main objective of this dissertation is to develop the second generation of Distributed Digital Radio (DDR) technology. A DDR II modem provides an integrated voice/data service platform, higher data rates and better throughput performance as compared to a DDR I modem. In order to improve the physical layer performance of DDR modems an analytical framework is first developed to model the Bit Error Rate (BER) performance of Orthogonal Frequency Division Multiplexing over Frequency Modulation (OFDM/FM) systems. The use of OFDM provides a spectrally efficient method of transmitting data over LMR channels. However, the high Peak-to-Average (PAR) of OFDM signals results in either a low Signal-to-Noise Ratio (SNR) at FM receiver or a high non-linear distortion of baseband signal in the FM transmitter. This dissertation presents an analytical framework to highlight the impact of high PAR of OFDM signal on OFDM/FM systems. A novel technique for reduction of PAR of OFDM called Linear Scaling Technique (LST) is developed. The use of LST mitigates the signal distortion occurring in OFDM over FM systems. Another important factor which affects the throughput of LMR networks is the Push-to-Talk (PTT) delay. A PTT delay refers to the delay between the instant when a PTT switch on a conventional LMR radio is keyed/unkeyed and a response is observed at the radio output. It can be separated into a Receive-To-Transmit Switch Interval (RTSI) or a Transmit-To-Receive Switch Interval (TRSI). This dissertation presents the typical RTSI delay values, distributions and their impact on throughput performance of LMR networks. An analytical model is developed to highlight the asymmetric throughput problem and the unintentional denial of service (UDOS) occurring in heterogeneous LMR networks consisting of radios with different PTT delay profiles. This information will be useful in performance and capacity planning of LMR networks in future

Peak to average power ratio reduction and error control in MIMO-OFDM HARQ System

Currently, multiple-input multiple-output orthogonal frequency division multiplexing (MIMOOFDM) systems underlie crucial wireless communication systems such as commercial 4G and 5G networks, tactical communication, and interoperable Public Safety communications. However, one drawback arising from OFDM modulation is its resulting high peak-to-average power ratio (PAPR). This problem increases with an increase in the number of transmit antennas. In this work, a new hybrid PAPR reduction technique is proposed for space-time block coding (STBC) MIMO-OFDM systems that combine the coding capabilities to PAPR reduction methods, while leveraging the new degree of freedom provided by the presence of multiple transmit chairs (MIMO). In the first part, we presented an extensive literature review of PAPR reduction techniques for OFDM and MIMO-OFDM systems. The work developed a PAPR reduction technique taxonomy, and analyzed the motivations for reducing the PAPR in current communication systems, emphasizing two important motivations such as power savings and coverage gain. In the tax onomy presented here, we include a new category, namely, hybrid techniques. Additionally, we drew a conclusion regarding the importance of hybrid PAPR reduction techniques. In the second part, we studied the effect of forward error correction (FEC) codes on the PAPR for the coded OFDM (COFDM) system. We simulated and compared the CCDF of the PAPR and its relationship with the autocorrelation of the COFDM signal before the inverse fast Fourier transform (IFFT) block. This allows to conclude on the main characteristics of the codes that generate high peaks in the COFDM signal, and therefore, the optimal parameters in order to reduce PAPR. We emphasize our study in FEC codes as linear block codes, and convolutional codes. Finally, we proposed a new hybrid PAPR reduction technique for an STBC MIMO-OFDM system, in which the convolutional code is optimized to avoid PAPR degradation, which also combines successive suboptimal cross-antenna rotation and inversion (SS-CARI) and iterative modified companding and filtering schemes. The new method permits to obtain a significant net gain for the system, i.e., considerable PAPR reduction, bit error rate (BER) gain as compared to the basic MIMO-OFDM system, low complexity, and reduced spectral splatter. The new hybrid technique was extensively evaluated by simulation, and the complementary cumulative distribution function (CCDF), the BER, and the power spectral density (PSD) were compared to the original STBC MIMO-OFDM signal

A study and experiment plan for digital mobile communication via satellite

The viability of mobile communications is examined within the context of a frequency division multiple access, single channel per carrier satellite system emphasizing digital techniques to serve a large population of users. The intent is to provide the mobile users with a grade of service consistant with the requirements for remote, rural (perhaps emergency) voice communications, but which approaches toll quality speech. A traffic model is derived on which to base the determination of the required maximum number of satellite channels to provide the anticipated level of service. Various voice digitalization and digital modulation schemes are reviewed along with a general link analysis of the mobile system. Demand assignment multiple access considerations and analysis tradeoffs are presented. Finally, a completed configuration is described

Systèmes OFDM optiques à détection directe à complexité réduite pour les communications à haut débit

A possible approach to maximize the data rate per wavelength, is to employ the high spectral efficiency discrete multitone (DMT) modulation. The work presented in this thesis mainly focuses on optimizing the power consumption and cost of DMT, that are the major obstacles to its market development. Within this context, we have first developed novel techniques permitting to discard the use of Hermitian symmetry in DMT modulations, thus significantly reducing the power consumption and the system cost. We have next proposed an asymmetric linear companding algorithm permitting to reduce the optical power of conventional DCO-OFDM modulation with a moderate complexity. A new VCSEL behavioural model based on the use of the VCSEL quasi-static characteristic was also developed to accurately evaluate the VCSEL impact on DMT modulations. Finally, we have built an experimental system to experimentally validate our proposed techniques. Several simulations and measurement results are then provided.Une approche pour augmenter le débit par longueur d'onde, est d'utiliser la modulation DMT (Discrete Multitone) à haute efficacité spectrale. Le travail présenté dans cette thèse se focalise principalement sur l'optimisation de la consommation en puissance et le coût de la DMT, qui présentent des obstacles majeurs à son industrialisation. Dans ce cadre, nous avons tout d'abord développé des nouvelles techniques permettant d'exclure la symétrie Hermitienne des modulations DMT, réduisant ainsi considérablement la consommation en puissance et le coût du système. Nous avons ensuite proposé un algorithme de compression linéaire asymétrique permettant de réduire la puissance optique de la modulation DMT avec une complexité modérée. Un nouveau modèle comportemental du VCSEL basé sur la caractéristique quasi-statique a été également développé. Nous avons enfin validé expérimentalement les techniques que nous avons proposées. Plusieurs résultats de simulations et de mesures sont ainsi présentés