289 research outputs found

    Speech coding at medium bit rates using analysis by synthesis techniques

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    Speech coding at medium bit rates using analysis by synthesis technique

    Novel Pitch Detection Algorithm With Application to Speech Coding

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    This thesis introduces a novel method for accurate pitch detection and speech segmentation, named Multi-feature, Autocorrelation (ACR) and Wavelet Technique (MAWT). MAWT uses feature extraction, and ACR applied on Linear Predictive Coding (LPC) residuals, with a wavelet-based refinement step. MAWT opens the way for a unique approach to modeling: although speech is divided into segments, the success of voicing decisions is not crucial. Experiments demonstrate the superiority of MAWT in pitch period detection accuracy over existing methods, and illustrate its advantages for speech segmentation. These advantages are more pronounced for gain-varying and transitional speech, and under noisy conditions

    Perceptual models in speech quality assessment and coding

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    The ever-increasing demand for good communications/toll quality speech has created a renewed interest into the perceptual impact of rate compression. Two general areas are investigated in this work, namely speech quality assessment and speech coding. In the field of speech quality assessment, a model is developed which simulates the processing stages of the peripheral auditory system. At the output of the model a "running" auditory spectrum is obtained. This represents the auditory (spectral) equivalent of any acoustic sound such as speech. Auditory spectra from coded speech segments serve as inputs to a second model. This model simulates the information centre in the brain which performs the speech quality assessment. [Continues.

    Coding of speech and image signals using Gabor decomposition

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    Ankara : The Department of Electrical and Electronics Engineering and the Institute of Engineerig and Sciences of Bilkent Univ., 1994.Thesis (Master's) -- Bilkent University, 1994.Includes bibliographical references leaves 27-28.A new low bit rate speech coding method which uses Gabor time-frequency decomposition and the matching pursuit algorithm is developed. A new algorithm based on the projections onto convex sets method is used to smooth the discontinuities between speech frames. A two-dimensional extension of the Gabor time-frequency decomposition is also developed for image coding. Simulation examples are presented.Gündüzhan, EmreM.S

    New Directions in Subband Coding

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    Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processo

    Speech coding at 4800 bps for mobile satellite communications

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    A speech compression project has recently been completed to develop a speech coding algorithm suitable for operation in a mobile satellite environment aimed at providing telephone quality natural speech at 4.8 kbps. The work has resulted in two alternative techniques which achieve reasonably good communications quality at 4.8 kbps while tolerating vehicle noise and rather severe channel impairments. The algorithms are embodied in a compact self-contained prototype consisting of two AT and T 32-bit floating-point DSP32 digital signal processors (DSP). A Motorola 68HC11 microcomputer chip serves as the board controller and interface handler. On a wirewrapped card, the prototype's circuit footprint amounts to only 200 sq cm, and consumes about 9 watts of power

    Implementing Linear Predictive Coding based on a statistical model for LTE fronthaul

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    This thesis studies the application of Linear Predictive coding (LPC) in the downlink of Long Term Evolution (LTE) fronthaul, which comprises of BBU and RRH. This can act as an additional module in the existing system. Today, the transmission of a single complex sample from the BBU to the RRH consumes 30 bits. The research of the thesis is to analyze the application of linear prediction theory in the LTE downlink transmission, which will work as a compression scheme and reduce this 30 bits to lower value, at the same time fulfill the Error Vector Magnitude (EVM) requirement stated in the LTE standards made by 3rd Generation Partnership Project (3GPP). As 4G-LTE and the upcoming access technologies will deal with large number of data samples in the transmission, it is an advantage if those data samples can be compressed without destroying the information content. LPC or linear prediction coding has been proved to be a very effective method for speech compression in audio related applications. In this thesis, the same logic of compression is applied on digital data samples of the LTE and the results are analyzed. It is found that, if LPC is applied properly on the LTE, it is possible to compress data samples efficiently and transmit them from the BBU to the RRH with fewer bits. At the RRH those compressed data samples can be processed and the main information data can be reconstructed, with additional quantization error and noise. This is obvious because LPC is a lossy compression method. A statistical model is established to generate a table of linear prediction filter coefficients which will be present both at the BBU and the RRH, when compression and decompression of data samples are performed. Entropy is also calculated in order to analyze the achievable compression on an actual error vector after implementing certain compression coding such as Huffman coding. The specific coding technique is left as a scope of future research.Due to the growth of number of users and faster communication methods, mobile operators have to use the allocated resources more efficiently to meet the user demands. Like any other systems, mobile communication networks go through series of updates over time. In mobile communication system, these updates are known as “Releases”. The transition from 3rd Generation (3G) to 4th Generation (4G) took place with Release 8 in 2008. Many new techniques are introduced in 4G in order to use the available resources more efficiently for improving quality of services (QoSs). LTE (Long term evolution) or more commonly known as 4G communication system deals with much larger amount of data traffic than any other previous technologies. Hence it is of utmost importance that the operators make use of the allocated bandwidth more efficiently to serve the ever increasing number of users. It is possible for LTE to deal with this large amount of data due to the use of OFDM modulation technique which ensures better quality of communication. In OFDM, there exists multiple blocks of frequency bands stacked together as a whole, which are not related to one another. The LTE structure is different from any previous systems. In telecommunication systems, there exists a unit which handles all the data traffic to and from the transmitter and the receiver. This module is called the base station. In LTE, the base station is divided into two parts namely the Baseband Unit (BBU) and the Radio Unit (RU), where almost all the data processing takes place at the BBU, and the RU is used as both transmitter and receiver when data is exchanged to and from a mobile device. In recent years, a new type of architecture is proposed, which is called the C-RAN (Cloud Radio Access Network). In C-RAN, the BBU and the RU would be placed at two different locations. Multiple BBUs can be placed together at a single place called the BBU pool, whereas the RUs will be placed in separate places far from the BBU pool and connected via optical fibers. In this structure, RU is known as RRH (Remote Radio Head) as they are separated from the BBU. One main advantage of such a structure is that, only the RRH is placed near the user locality and the BBU can be put at the network operator’s vicinity. This also helps in reducing the operating and maintenance cost for the operator in many ways. Since the LTE imposes with massive amount of data traffic on the fronthaul (almost tenfold of the actual information data after applying error correcting coding, control signals etc.), it is very important to carry out compression of those data traffic before they are sent from the BBU to the RU. If good compression is carried out, then it becomes possible to accommodate more users, using the available resources. Although analog signals are used to transmit a message from the transmitter to the receiver over a medium, it is always important to convert those analog signal to digital signal to be transmitted from one block to the next block for processing, through the connecting link. The main purpose of this thesis work is to apply a compression technique which will minimize the number of bits needed to represent each of those data samples transmitted from the BBU to the RRH. The compression technique used in this thesis is to employ a module which will use certain number of previous data samples values to make a prediction of the next data sample. Then this predicted data sample is compared with the actual data sample and their difference is found. The difference between these two samples has a low magnitude, as a result it is possible to use lower number of bits in the digital domain to represent this value, and finally transmitted through the link to the RRH. At the RRH, the same prediction module is used to utilize these received samples of low magnitude, to make a prediction of the original data samples which are intended to be sent at the first place. In order to make the prediction module to function properly, it is very important to set up the filter values, which are known as the prediction coefficients. These coefficients play the role of successfully predicting data samples which are very similar to the original data samples. These coefficients are calculated by statistical method so that they can be used for any set of random data sample vector in the LTE. This thesis studies the performance of applying this prediction technique in LTE. In order to identify the efficiency of this applied compression technique, certain parameters are calculated using various simulations, and compared with the set of values as specified by the main researching bodies of the LTE. It is found that, the applied compression technique works fine in LTE as the simulation results support the validity of the scheme. It also proves that, it is possible to introduce this compression technique as an extension to the upcoming upgrades of the LTE, and this will facilitate accommodating more users with the available infrastructure resources

    Solid rocket motor internal insulation

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    Internal insulation in a solid rocket motor is defined as a layer of heat barrier material placed between the internal surface of the case propellant. The primary purpose is to prevent the case from reaching temperatures that endanger its structural integrity. Secondary functions of the insulation are listed and guidelines for avoiding critical problems in the development of internal insulation for rocket motors are presented
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