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Abstract The main objective of this paper is to explore parameters usually adopted by the JPEG method and use them in speech signal compression. Speech compression is the technique of encoding the speech signal in some way that allows the same speech parameters to represent the whole signal. In other words, it is to eliminate redundant features of speech and keep only the important ones for the next stage of speech reproduction. In this paper, the proposed method is to adopt the JPEG scheme, which is usually used in digital image compression and digital speech signal compression. This will open the door to use some methods that are already designed to fit two-dimensional (2D) signals and use them in a 1D signal world. The method includes many priori steps for preparing the speech signal to make it more comparable with the JPEG technique. In order to examine the quality of the compressed data using the JPEG method, different quantization matrices with different compression levels have been employed within the proposed method. Comparison results between the original signal and the reproduced (decompressed) signal show a huge matching. However, different quantization matrices could reduce the quality of the reproduced signal, but in general, it is still within the range of acceptance.

Recently, data of multimedia performs an exponentially blowing tendency, saturating daily life of humans. Various modalities of data, includes images, texts and video, plays important role in different aspects and has wide. However, the key problem of utilizing data of large scale is cost of processing and massive storage. Therefore, for efficient communications and for economical storage requires effective techniques of data compression to reduce the volume of data. Speech coding is a main problem in the area of digital speech processing. The process of converting the voice signals into a more compressed form is speech coding. In this work, we demonstrate that a DCT with a chaotic system combined with run-length coding can be utilized to implement speech coding of very low bit-rate with high reconstruction quality. Experimental result show that compression ratio is about 13% when implemented on Librispeech dataset.

In this paper, an updated source-filter based method is used for speech recognition of English speakers. The key goal of this paper is to discuss the influence of using dynamic and selective Filterbank on the efficiency of the system in terms of identifying the word being said. Dynamic structures of the bandpass filter bank are clearly able to adjust their principles in conjunction with the speech database used. In this paper, the proposed methods adopted the idea of re-select or adjusting the filter parameters values at each iteration, in accordance with the word being processed. In other words, the values of the filter components (center frequencies and bandwidths) will adjust until it gets the best set of the filter components. The genetic algorithm (GA) will be used to maintain the components of the filter and each new population of the GA represents a new set of channels for the filter bank. Results show that the efficiency of the recognition system would be improved with each new set of filter components generated by the genetic algorithm. Speech recognition, Source-filter, Filterbank, genetic algorithm, adjusted filter, Dynamic system

OWC systems have turned into a state-of-the-art technology because of their superior performance uniqueness and its innumerable characteristic as compared to its RF. One of the OWC application is inter-satellite wireless optical communication (Is-WOC), this framework can be improved by using different ways such as advanced modulation formats and aperture diameter. We demonstrate an analysis based on the aperture diameter effect in transmission of single channel with 40 Gb/s Is-WOC system using three different modulation methods, differential phase-shift keying (DPSK) with a duty ratio to 66% and 33% and duobinary. Choosing these three modulations from various modulation format is done depending on the advantage they provide where the DPSK gives to fiber nonlinearity a powerful robustness and less susceptible to SPM and XPM and for system requirements of dispersion compensation in duobinary, it provides higher chromatic dispersion tolerance mitigate. The result shows the effect of different transmitter aperture diameter (from 2 to 20 cm) for a constant distance 200 km accomplished by quality factor and minimum BER. We conclude there is a great loss in the small aperture diameter, even in the presence of best modulation format, therefore as the aperture diameter increase the Q-factor increase, but some increased rise is linear and the other is changing from linear to non-linear in a fixed Q-factor point equal to 5.63.