Adaptive error protection coding for wireless transmission of motion JPEG 2000 video

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GCL Based Synchronization and Time Domain Frequency Offset Correction in OFDM System

Orthogonal Frequency Division Multiplexing (OFDM) is a modulation technique that has become the technology of choice in most wireless communication networks of today. Despite the advantages the OFDM system offers, it has some disadvantages like sensitivity to synchronization and high power-to-average-power ratio (PAPR). Any time offset leads to inter-symbol interference (ISI) whereas any frequency offset results in inter-carrier interference (ICI) and, as a result, the system performance degrades. The studies of preamble based time synchronization show that, the standard PN sequence based preamble in IEEE 802.16a is less robust to frequency offset when used in Park’s method of time synchronization - a method that gives more accurate result as compared to other methods. Time domain channel estimation cannot be carried out in the presence of integer frequency offset. This thesis has three specific objectives. Firstly, to design and evaluate a new preamble by making use of a generalized chirp-like (GCL) sequence that has low PAPR and good time and also frequency correlation properties. Secondly, to design a new receiver scheme that estimates and corrects the integer-frequency offset in the time domain and evaluate its performance. And lastly, having corrected the frequency offset in time domain, to estimate the wireless channel in time domain and evaluate its performance. The results show that, the proposed GCL based preamble has better and more robust performance than the standard PN sequence (IEEE 802.16 standard) based preamble in the time and integer frequency synchronization and also in the time domain channel estimation. In the new receiver scheme, the presence of symmetrical correlation shows that received signal is frequency corrected. The results show that the new receiver scheme is able to detect the symmetrical correlation quite accurately. The receiver also works well even in low SNR environment

Progressive transmission of medical images

A novel adaptive source-channel coding scheme for progressive transmission of medical images with a feedback system is therefore proposed in this dissertation. The overall design includes Discrete Wavelet Transform (DWT), Embedded Zerotree Wavelet (EZW) coding, Joint Source-Channel Coding (JSCC), prioritization of region of interest (RoI), variability of parity length based on feedback, and the corresponding hardware design utilising Simulink. The JSCC can achieve an efficient transmission by incorporating unequal error projection (UEP) and rate allocation. An algorithm is also developed to estimate the number of erroneous data in the receiver. The algorithm detects the address in which the number of symbols for each subblock is indicated, and reassigns an estimated correct data according to a decision making criterion, if error data is detected. The proposed system has been designed based on Simulink which can be used to generate netlist for portable devices. A new compression method called Compressive Sensing (CS) is also revisited in this work. CS exhibits many advantages in comparison with EZW based on our experimental results. DICOM JPEG2000 is an efficient coding standard for lossy or lossless multi-component image coding. However, it does not provide any mechanism for automatic RoI definition, and is more complex compared to our proposed scheme. The proposed system significantly reduces the transmission time, lowers computation cost, and maintains an error-free state in the RoI with regards to the above provided features. A MATLAB-based TCP/IP connection is established to demonstrate the efficacy of the proposed interactive and adaptive progressive transmission system. The proposed system is simulated for both binary and symmetric channel (BSC) and Rayleigh channel. The experimental results confirm the effectiveness of the design.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Layer-based coding, smoothing, and scheduling of low-bit-rate video for teleconferencing over tactical ATM networks

This work investigates issues related to distribution of low bit rate video within the context of a teleconferencing application deployed over a tactical ATM network. The main objective is to develop mechanisms that support transmission of low bit rate video streams as a series of scalable layers that progressively improve quality. The hierarchical nature of the layered video stream is actively exploited along the transmission path from the sender to the recipients to facilitate transmission. A new layered coder design tailored to video teleconferencing in the tactical environment is proposed. Macroblocks selected due to scene motion are layered via subband decomposition using the fast Haar transform. A generalized layering scheme groups the subbands to form an arbitrary number of layers. As a layering scheme suitable for low motion video is unsuitable for static slides, the coder adapts the layering scheme to the video content. A suboptimal rate control mechanism that reduces the kappa dimensional rate distortion problem resulting from the use of multiple quantizers tailored to each layer to a 1 dimensional problem by creating a single rate distortion curve for the coder in terms of a suboptimal set of kappa dimensional quantizer vectors is investigated. Rate control is thus simplified into a table lookup of a codebook containing the suboptimal quantizer vectors. The rate controller is ideal for real time video and limits fluctuations in the bit stream with no corresponding visible fluctuations in perceptual quality. A traffic smoother prior to network entry is developed to increase queuing and scheduler efficiency. Three levels of smoothing are studied: frame, layer, and cell interarrival. Frame level smoothing occurs via rate control at the application. Interleaving and cell interarrival smoothing are accomplished using a leaky bucket mechanism inserted prior to the adaptation layer or within the adaptation layerhttp://www.archive.org/details/layerbasedcoding00parkLieutenant Commander, United States NavyApproved for public release; distribution is unlimited

Speech Recognition

Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes