Dissertations / Theses: 'Bit rate'

1

Kritzinger, Carl. "Low bit rate speech coding." Thesis, Stellenbosch : University of Stellenbosch, 2006. http://hdl.handle.net/10019.1/2078.

Full text

Abstract:

Thesis (MScIng (Electrical and Electronic Engineering))--University of Stellenbosch, 2006.
Despite enormous advances in digital communication, the voice is still the primary tool with which people exchange ideas. However, uncompressed digital speech tends to require prohibitively high data rates (upward of 64kbps), making it impractical for many applications. Speech coding is the process of reducing the data rate of digital voice to manageable levels. Parametric speech coders or vocoders utilise a-priori information about the mechanism by which speech is produced in order to achieve extremely efficient compression of speech signals (as low as 1 kbps). The greater part of this thesis comprises an investigation into parametric speech coding. This consisted of a review of the mathematical and heuristic tools used in parametric speech coding, as well as the implementation of an accepted standard algorithm for parametric voice coding. In order to examine avenues of improvement for the existing vocoders, we examined some of the mathematical structure underlying parametric speech coding. Following on from this, we developed a novel approach to parametric speech coding which obtained promising results under both objective and subjective evaluation. An additional contribution by this thesis was the comparative subjective evaluation of the effect of parametric speech coding on English and Xhosa speech. We investigated the performance of two different encoding algorithms on the two languages.

APA, Harvard, Vancouver, ISO, and other styles

2

Guadiana, Juan M., and Fil Macias. "ENCRYPTED BIT ERROR RATE TESTING." International Foundation for Telemetering, 2002. http://hdl.handle.net/10150/607507.

Full text

Abstract:

International Telemetering Conference Proceedings / October 21, 2002 / Town & Country Hotel and Conference Center, San Diego, California
End-to-End testing is a tool for verifying that Range Telemetry (TM) System Equipment will deliver satisfactory performance throughout a planned flight test. A thorough test verifies system thresholds while gauging projected mission loading all in the presence of expected interference. At the White Sands Missile Range (WSMR) in New Mexico, system tests are routinely conducted by Range telemetry Engineers and technicians in the interest of ensuring highly reliable telemetry acquisition. Even so, flight or integration tests are occasionally halted, unable to complete these telemetry checks. The Navy Standard Missile Program Office and the White Sands Missile Range, have proactively conducted investigations to identify and eliminate problems. A background discussion is provided on the serious problems with the launcher acquisition, which were resolved along the way laying the ground work for effective system testing. Since there were no provisions to test with the decryption equipment an assumption must be made. Encryption is operationally transparent and reliable. Encryption has wide application, and for that reason the above assumption must be made with confidence. A comprehensive mission day encrypted systems test is proposed. Those involved with encrypted telemetry systems, and those experiencing seemingly unexplainable data degradations and other problems with or without encryption should review this information.

APA, Harvard, Vancouver, ISO, and other styles

3

Plain, Simon E. M. "Bit rate scalability in audio coding." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0034/MQ64243.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

4

Bicket, John C. (John Charles). "Bit-rate selection in wireless networks." Thesis, Massachusetts Institute of Technology, 2005. http://hdl.handle.net/1721.1/34116.

Full text

Abstract:

Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2005.
Includes bibliographical references (p. 49-50).
This thesis evaluates bit-rate selection techniques to maximize throughput over wireless links that are capable of multiple bit-rates. The key challenges in bit-rate selection are determining which bit-rate provides the most throughput and knowing when to switch to another bit-rate that would provide more throughput. This thesis presents the SampleRate bit-rate selection algorithm. SampleRate sends most data packets at the bit-rate it believes will provide the highest throughput. SampleRate periodically sends a data packet at some other bit-rate in order to update a record of that bit-rate's loss rate. SampleRate switches to a different bit-rate if the throughput estimate based on the other bit-rate's recorded loss rate is higher than the current bit-rate's throughput. Measuring the loss rate of each supported bit-rate would be inefficient because sending packets at lower bit-rates could waste transmission time, and because successive unicast losses are time-consuming for bit-rates that do not work. SampleRate addresses this problem by only sampling at bit-rates whose lossless throughput is better than the current bit-rate's throughput. SampleRate also stops probing at a bit-rate if it experiences several successive losses. This thesis presents measurements from indoor and outdoor wireless networks that demonstrate that SampleRate performs as well or better than other bit-rate selection algorithms.
(cont.) SampleRate performs better than other algorithms on links where all bit-rates suffer from significant loss.
by John C. Bicket.
S.M.

APA, Harvard, Vancouver, ISO, and other styles

5

Moen, Selmer, and Charles Jones. "BIT RATE AGILITY FOR EFFICIENT TELEMETRY." International Foundation for Telemetering, 2003. http://hdl.handle.net/10150/606754.

Full text

Abstract:

International Telemetering Conference Proceedings / October 20-23, 2003 / Riviera Hotel and Convention Center, Las Vegas, Nevada
The Bit Rate Agile Onboard Telemetry Formatting (BRAOTF) system was developed by Killdeer Mountain Manufacturing to address increasing demands on the efficiency of telemetry systems. The BRAOTF thins and reorders data streams, adjusting the bit rate of a pulse code modulation (PCM) stream using a bit-locked loop to match the desired information rate exactly. The BRAOTF accomplishes the adjustment in hardware, synthesizing a clock whose operating frequency is derived from the actual timing of the input format. Its firmware manages initialization and error management. Testing has confirmed that the BRAOTF implementation meets its design goals.

APA, Harvard, Vancouver, ISO, and other styles

6

Cubiss, Christopher. "Low bit-rate image sequence coding." Thesis, University of Edinburgh, 1994. http://hdl.handle.net/1842/13506.

Full text

Abstract:

Digital video, by its very nature, contains vast amounts of data. Indeed, the storage and transmission requirements of digital video frequency far exceed practical storage and transmission capacity. Therefore such research has been dedicated to developing compression algorithms for digital video. This research has recently culminated in the introduction of several standards for image compression. The CCITT H.261 and the motion picture experts group (MPEG) standards both target full-motion video and are based upon a hybrid architecture which combines motion-compensated prediction with transform coding. Although motion-compensated transform coding has been shown to produce reasonable quality reconstructed images, it has also been shown that as the compression ratio is progressively increased the quality of the reconstructed image rapidly degrades. The reasons for this degradation are twofold: firstly, the transform coder is optimised for encoding real-world images, not prediction errors; and secondly, the motion-estimation and transform-coding algorithms both decompose the image into a regular array of blocks which, as the coding distortion is progressively increased, results in the well known 'blocking' effect. The regular structure of this coding artifact makes this error particularly disturbing. This research investigates motion estimation and motion compensated prediction with the aim of characterising the prediction error so that more optimal spatial coding algorithms can be chosen. Motion-compensated prediction was considered in detail. Simple theoretical models of the prediction error were developed and it was shown that, for sufficiently accurate motion estimates, motion-compensated prediction could be considered as a non-ideal spatial band-pass filtering operation. Rate-distortion theory was employed to show that the inverse spectral flatness measure of the prediction error provides a direct indication of the expected coding gain of an optimal hybrid motion-compensated prediction algorithm.

APA, Harvard, Vancouver, ISO, and other styles

7

CHAKRABORTY, S. K., and R. K. RAJANGAM. "PROGRAMMABLE HIGH BIT RATE FRAME SYNCHRONISER." International Foundation for Telemetering, 1989. http://hdl.handle.net/10150/614490.

Full text

Abstract:

International Telemetering Conference Proceedings / October 30-November 02, 1989 / Town & Country Hotel & Convention Center, San Diego, California
The first Indian Remote Sensing Satellite was launched on 17th March 1988 from a Soviet Cosmodrome into a 904 Km Polar Sunsynchronous orbit. The data transmission from the satellite is at 5.2 Mega Bits/sec in S-Band and 10.4 Mega Bits/sec in X-Band. The payload data is formatted into custom made 8328 words format. A programmable unique versatile frame sync and Decommutation unit has been developed to test the data from the data handling system during its various phases of development. The system works upto 50 Mega Bits/sec and can handle frame sync code length upto 128 bits and a frame length of 2 Exp 20 bits. Provision has been made for programming the allowable bit errors as well as bit slippages, using a front panel setting. This paper describes the design and implementation of such a high bit rate frame synchroniser developed specially for IRS Spacecraft application. It will also highlight the performance of the system.

APA, Harvard, Vancouver, ISO, and other styles

8

吳景濤 and King-to Ng. "A novel bit allocation buffer control algorithm for low bit-rate videocompression." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1998. http://hub.hku.hk/bib/B31221518.

Full text

APA, Harvard, Vancouver, ISO, and other styles

9

Saw, Yoo-Sok. "Nonlinear rate control techniques for constant bit rate MPEG video coders." Thesis, University of Edinburgh, 1997. http://hdl.handle.net/1842/1381.

Full text

Abstract:

Digital visual communication has been increasingly adopted as an efficient new medium in a variety of different fields; multi-media computers, digital televisions, telecommunications, etc. Exchange of visual information between remote sites requires that digital video is encoded by compressing the amount of data and transmitting it through specified network connections. The compression and transmission of digital video is an amalgamation of statistical data coding processes, which aims at efficient exchange of visual information without technical barriers due to different standards, services, media, etc. It is associated with a series of different disciplines of digital signal processing, each of which can be applied independently. It includes a few different technical principles; distortionrate theory, prediction techniques and control theory. The MPEG (Moving Picture Experts Group) video compression standard is based on this paradigm, thus, it contains a variety of different coding parameters which may result in different performance depending on their values. It specifies the bit stream syntax and the decoding process as its normative parts. The encoder details remain nonnormative and are configured by a specific design. This means that the MPEG video encoder has a great deal of flexibility in the aspects of performance and implementation. This thesis deals with control techniques for the data rate of compressed video, which determine the encoding efficiency and video quality. The video rate control is achieved by adjusting quantisation step size depending on the occupancy of a transmission buffer memory which stores the compressed video data for a specific period of time. Conventional video rate control techniques have generally been based either on linear predictive or on control theoretic models. However, this thesis takes a different view on digital video and MPEG video coding, and focuses on the non-stationary and nonlinear nature of realistic moving pictures. Furthermore, considering the MPEG encoding structure involved in the different disciplines, A series of improvements for video rate control are proposed, each of which enhances the performance of the MPEG encoder. A nonlinear quantisation control technique is investigated, which controls the buffer occupancy with the quantiser using a series of nonlinear functions. Linear and nonlinear feed-forward networks are also employed to control the quantiser. The linear combiner is used as a linear estimator and a radial basis function network as a nonlinear one. Finally, fuzzy rulebased control is applied to exploit the advantages of the nonlinear control technique which is able to provide linguistic judgement in the control mechanism. All these techniques are employed according to two global approaches (feedforward and feedback) applied to the rate control. The performance evaluation is carried out in terms of controllability over bit rate variation and video quality, by conducting a series of simulations.

APA, Harvard, Vancouver, ISO, and other styles

10

Ghani, Nasir. "Available bit rate services in ATM networks." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq22206.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

11

鄧世健 and Sai-kin Owen Tang. "Implementation of Low bit-rate image codec." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1994. http://hub.hku.hk/bib/B31212670.

Full text

APA, Harvard, Vancouver, ISO, and other styles

12

Scargall, Lee David. "Very low bit-rate digital video coding." Thesis, University of Newcastle Upon Tyne, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.299046.

Full text

APA, Harvard, Vancouver, ISO, and other styles

13

Gross, Samuel A. "Hardware implementation of wireless bit rate adaptation." Thesis, Massachusetts Institute of Technology, 2010. http://hdl.handle.net/1721.1/62641.

Full text

Abstract:

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2010.
Cataloged from PDF version of thesis.
Includes bibliographical references (p. 43-44).
This thesis presents a hardware implementation of the SoftRate bit-rate adaptation protocol. SoftRate is a new bit-rate adaptation protocol, which uses per-bit confidence hints generated by the convolutional decoder to estimate the channel bit-error rate. Implementing SoftRate requires changes to both the physical and media access control layers. which precludes using existing commodity 802.11 hardware. This project developed a SoftRate implementation on top of Airblue, an FPGA platform for developing wireless protocols. We present a hardware implementation of SoftRate which meets 802.11 timing requirements.
by Samuel A. Gross.
M.Eng.

APA, Harvard, Vancouver, ISO, and other styles

14

Griffin, Wesley. "Quality Guided Variable Bit Rate Texture Compression." Thesis, University of Maryland, Baltimore County, 2016. http://pqdtopen.proquest.com/#viewpdf?dispub=10159930.

Full text

Abstract:

The primary goal of computer graphics is to create images by rendering a scene under two constraints: quality, producing the image with as few artifacts as possible, and time, producing the image as fast as possible. Technology advances have both helped to satisfy these constraints, with Graphics Processing Unit (GPU) advances reducing image rendering times, and to exacerbate these constraints, with new HD and virtual reality displays increasing rendering resolutions. To meet both constraints, rendering uses texture mapping which maps 2D textures onto scene objects. Over time, the count and resolution of textures has increased, resulting in dramatic growth of data storage requirements. Compression can help to reduce these storage requirements.

I present a rigorous texture compression evaluation methodology using final rendered images. My method can account for masking effects introduced by the texture mapping process while leveraging the perceptual-rigor of current Image Quality Assessment metrics. Building on this evaluation methodology, I present a demonstration of guided texture compression optimization that minimizes the bitrate of compressed textures while maximizing the quality of final rendered images. Guided texture compression will help with the scalability problem for optimizing texture compression in real-world scenarios.

APA, Harvard, Vancouver, ISO, and other styles

15

Tang, Sai-kin Owen. "Implementation of Low bit-rate image codec /." [Hong Kong] : University of Hong Kong, 1994. http://sunzi.lib.hku.hk/hkuto/record.jsp?B14804402.

Full text

APA, Harvard, Vancouver, ISO, and other styles

16

Ng, King-to. "A novel bit allocation buffer control algorithm for low bit-rate video compression /." Hong Kong : University of Hong Kong, 1998. http://sunzi.lib.hku.hk/hkuto/record.jsp?B20192733.

Full text

APA, Harvard, Vancouver, ISO, and other styles

17

Chen, Wei 1976. "Perceptual postfiltering for low bit rate speech coders." Thesis, McGill University, 2007. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=112563.

Full text

Abstract:

Adaptive postfiltering has become a common part of speech coding standards based on the Linear Prediction Analysis-by-Synthesis algorithm to decrease audible coding noise. However, a conventional adaptive postfilter is based on empirical assumptions of masking phenomena, which sometimes makes it hard to balance between noise reduction and speech distortion.
This thesis introduces a novel perceptual postfiltering system for low bit rate speech coders. The proposed postfilter works at the decoder, as is the case for the conventional adaptive postfilter. Specific human auditory properties are considered in the postfilter design to improve speech quality. A Gaussian Mixture Model based Minimum Mean Squared Error estimation of the perceptual postfilter is performed with the received information at the decoder. Perceptual postfiltering is then applied to the reconstructed speech to improve speech quality. Test results show that the proposed system gives better perceptual speech quality over conventional adaptive postfiltering.

APA, Harvard, Vancouver, ISO, and other styles

18

Fadzil, Ahmad. "Video image sequence coding at low bit rate." Thesis, University of Essex, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.290612.

Full text

APA, Harvard, Vancouver, ISO, and other styles

19

Mustapha, Azhar K. 1975. "Postfiltering techniques in low bit-rate speech coders." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/80589.

Full text

Abstract:

Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes bibliographical references (leaves 78-80).
by Azhar K. Mustapha.
M.Eng.

APA, Harvard, Vancouver, ISO, and other styles

20

Nguyen, Ricky D. (Ricky Do). "Rate control and bit allocations for JPEG transcoding." Thesis, Massachusetts Institute of Technology, 2007. http://hdl.handle.net/1721.1/41667.

Full text

Abstract:

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2007.
Includes bibliographical references (leaves 50-51).
An image transcoder that produces a baseline JPEG file from a baseline JPEG input is developed. The goal is to produce a high quality image while accurately meeting a filesize target and keeping computational complexity-especially the memory usage and number of passes at the input image--low. Building upon the work of He and Mitra, the JPEG transcoder exploits a linear relationship between the number of zero-valued quantized DCT coefficients and the bitrate. Using this relationship and a histogram of coefficients, it is possible to determine an effective way to scale the quantization tables of an image to approach a target filesize. As the image is being transcoded, an intra-image process makes minor corrections, saving more bits as needed throughout the transcoding of the image. This intra-image process decrements specific coefficients, minimizing the change in value (and hence image quality) while maximizing the savings in bitrate. The result is a fast JPEG transcoder that reliably achieves a target filesize and preserves as much image quality as possible. The proposed transcoder and several variations were tested on a set of twenty-nine images that gave a fair representation of typical JPEG photos. The evaluation metric consisted of three parts: first, the accuracy and precision of the output filesize with respect to the target filesize; second, the PSNR of the output image with respect to the original image; and third, the subjective visual image quality.
by Ricky D. Nguyen.
M.Eng.

APA, Harvard, Vancouver, ISO, and other styles

21

Reed, Eric Christopher. "Multi-dimensional bit rate control for video communication." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/86669.

Full text

APA, Harvard, Vancouver, ISO, and other styles

22

Xianda, Dai, He Rongfa, and Li Man. "EFFECT OF CLOCK JITTERS ON BIT ERROR RATE." International Foundation for Telemetering, 1985. http://hdl.handle.net/10150/615762.

Full text

Abstract:

International Telemetering Conference Proceedings / October 28-31, 1985 / Riviera Hotel, Las Vegas, Nevada
In this paper we have formulated a method of analysing the effect of clock jitters, which, even with the monomode optical fiber avaiable today, is still there to limit the operational distance of a digital fiber optic system. The main intention is to compare the system performance of a conventional binary system with that of a newly developed four-level pulse width modulation (PWM) system. Calculated results show an improvement in combatting clock jitters when using the four-level PWM system.

APA, Harvard, Vancouver, ISO, and other styles

23

Sriram, Parthasarathy. "Low bit-rate transform and wavelet image coding." Diss., The University of Arizona, 1993. http://hdl.handle.net/10150/186365.

Full text

Abstract:

In this dissertation, adaptive wavelet and transform coding techniques are presented for low bit-rate monochrome and color image coding. The proposed encoders are based on trellis coded quantization. Trellis coded quantization (TCQ) is an effective scheme for quantizing memoryless sources with low to moderate complexity. The TCQ approach to data compression has led to some of the most effective source codes found to date for memoryless sources. For the transform coder, TCQ is used to encode transform coefficients resulting from applying a 16 x 16 discrete cosine transform (DCT) to 8-bit gray level and 24-bit color images. For the color images, the red, green, and the blue planes were transformed into NTSC transmission primaries (Y, I, and Q) before the DCT is applied. Both fixed-rate and entropy-constrained systems are considered. The discrete wavelet transform has recently emerged as a powerful technique for decomposing images into various multi-resolution approximations. We investigate the use of entropy-constrained trellis coded quantization for encoding the wavelet coefficients of both monochrome and color images. The lowest resolution sub-image is encoded using a 4 x 4 2-D DCT encoder. An integer programming algorithm is employed to allocate the available bit-rate optimally among the subbands. The objective performance results of our wavelet and transform coders are comparable to or surpass all previous results reported in the literature. The subjective quality of the encoded images is also excellent. In particular, the encoded monochrome images at 0.5 bits/pixel (a compression ratio of 16:1) obtained using our adaptive wavelet coder is almost indistinguishable from the original even when viewed on a high-resolution monitor.

APA, Harvard, Vancouver, ISO, and other styles

24

Moretti, Marc-Jules Bernard 1964. "Bit error rate computation in optical fiber communications." Thesis, The University of Arizona, 1991. http://hdl.handle.net/10150/278009.

Full text

Abstract:

The focus of this thesis is the computation of bit error rate in an optical fiber communication system when an avalanche photodiode (APD) is used for detection. The use of the APD increases the complexity of the BER calculation because it introduces an additional noise phenomenon: shot noise (plus multiplication noise). There are several methods to compute the BER. This thesis considers three of these methods: the Gaussian approximation, the saddlepoint approximation and a numerical quadrature method. An in-depth analysis of each computation method is presented after a thorough study of APD characteristics, involving its probability density function and its moment generating function. Numerical examples are shown and compared for each method. The examples show that the saddlepoint approximation method can be used to provide an accurate, simple form for the bit error rate, and that the Gaussian approximation tends to underestimate the BER at high gains. Different results with different photodetectors are illustrated throughout the thesis for a better understanding.

APA, Harvard, Vancouver, ISO, and other styles

25

Wyman, Richard Hayden. "Bit-plane differential EZW for the compression of video for available bit-rate channels." Thesis, Imperial College London, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.313533.

Full text

APA, Harvard, Vancouver, ISO, and other styles

26

Schmidt-Langhorst, Carsten. "Optical sampling of high bit rate optical data signals." [S.l.] : [s.n.], 2004. http://deposit.ddb.de/cgi-bin/dokserv?idn=971438358.

Full text

APA, Harvard, Vancouver, ISO, and other styles

27

Unver, Emre. "Advanced Low Bit-Rate Speech Coding Below 2.4 Kbps." Thesis, University of Surrey, 2010. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.518687.

Full text

APA, Harvard, Vancouver, ISO, and other styles

28

Yaghmaie, Khashayar. "Prototype waveform interpolation based low bit rate speech coding." Thesis, University of Surrey, 1997. http://epubs.surrey.ac.uk/843152/.

Full text

Abstract:

Advances in digital technology in the last decade have motivated the development of very efficient and high quality speech compression algorithms. While in the early low bit rate coding systems, the main target was production of intelligible speech at low information rates, expansion of new applications such as mobile satellite systems increased the demand for high quality speech at lowest possible bit rates. This resulted in the development of efficient parametric models for speech production system. These models were the basis of powerful speech compression algorithms such as CELP and Multiband excitation. CELP is a very efficient algorithm at medium bit rates and has achieved almost toll quality at 8 kb/s. However, the performance of CELP rapidly reduces at bit rates below 4.8 kb/s. The sinusoidal based coding algorithms and in particular multiband excitation technique have proved their abilities in producing high quality speech at bit rates below 5 kb/s. In recent years, another efficient speech compression algorithm called prototype waveform interpolation (PWI) has emerged. PWI presented a novel model which proved to be very efficient in removing redundant information from speech. While the early PWI systems produced high quality speech at bit rates around 3.5 kb/s, its latest versions produce an even higher quality at the bit rates as low as 2.4 kb/s. The key to the success of PWI is the approach it exploits in reducing the distortion associated with low bit rate coding algorithms. However, the price for this achievement is a very high computational demand which has been the main hurdle in its real time applications. The aim of the research in this thesis is the development of low complexity PWI systems without sacrificing the high quality. While the target of the majority of PWI systems is efficient coding of the excitation signal in the LP model of speech, this research focuses on exploiting PWI to directly encode the original speech. In the first part of the thesis, basic techniques in low bit rate speech coding are described and proper tools are developed to be exploited in a PWI based coding system. In the second part, the original PWI algorithm operating in the LP residual domain is briefly explained and application of PWI in speech domain is introduced as a method to cope with problems associated with the original PWI. To demonstrate the abilities of this approach, various coding schemes operating in the range of 1.85 to 2.95 kb/s are developed. In the final stage, a new technique which combines the two powerful low bit rate coding techniques, i.e multiband excitation and PWI, is developed to produce high quality synthetic speech at 2.6 kb/s.

APA, Harvard, Vancouver, ISO, and other styles

29

Lee, Il-Sung. "Traffic shaping for variable-bit-rate MPEG-2 video." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp01/MQ37266.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

30

Papacostantinou, Costantinos. "Improved pitch modelling for low bit-rate speech coders." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp01/MQ37279.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

31

Alexopoulos, Kyriakos. "Phase spectral representation for low bit rate speech coding." Thesis, Imperial College London, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.249314.

Full text

APA, Harvard, Vancouver, ISO, and other styles

32

Easton, Mark. "Fast algorithms for low bit rate digital speech coding." Thesis, University of Liverpool, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.240500.

Full text

APA, Harvard, Vancouver, ISO, and other styles

33

Edirisinghe, Sumuda G. "Dispersion compensation techniques in high bit rate transmission systems." Thesis, University of Essex, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.343578.

Full text

APA, Harvard, Vancouver, ISO, and other styles

34

Nakhai, Mohammad Reza. "A low bit rate speech codec for wireless applications." Thesis, King's College London (University of London), 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.392144.

Full text

APA, Harvard, Vancouver, ISO, and other styles

35

Brooks, Fiona Clare Angharad. "Very low bit rate voice compression for mobile communications." Thesis, University of Southampton, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.287341.

Full text

APA, Harvard, Vancouver, ISO, and other styles

36

Kweh, Teck Hock. "Improved quality block-based low bit rate video coding." Thesis, University of Surrey, 1998. http://epubs.surrey.ac.uk/844563/.

Full text

Abstract:

The aim of this research is to develop algorithms for enhancing the subjective quality and coding efficiency of standard block-based video coders. In the past few years, numerous video coding standards based on motion-compensated block-transform structure have been established where block-based motion estimation is used for reducing the correlation between consecutive images and block transform is used for coding the resulting motion-compensated residual images. Due to the use of predictive differential coding and variable length coding techniques, the output data rate exhibits extreme fluctuations. A rate control algorithm is devised for achieving a stable output data rate. This rate control algorithm, which is essentially a bit-rate estimation algorithm, is then employed in a bit-allocation algorithm for improving the visual quality of the coded images, based on some prior knowledge of the images. Block-based hybrid coders achieve high compression ratio mainly due to the employment of a motion estimation and compensation stage in the coding process. The conventional bit-allocation strategy for these coders simply assigns the bits required by the motion vectors and the rest to the residual image. However, at very low bit-rates, this bit-allocation strategy is inadequate as the motion vector bits takes up a considerable portion of the total bit-rate. A rate-constrained selection algorithm is presented where an analysis-by-synthesis approach is used for choosing the best motion vectors in term of resulting bit rate and image quality. This selection algorithm is then implemented for mode selection. A simple algorithm based on the above-mentioned bit-rate estimation algorithm is developed for the latter to reduce the computational complexity. For very low bit-rate applications, it is well-known that block-based coders suffer from blocking artifacts. A coding mode is presented for reducing these annoying artifacts by coding a down-sampled version of the residual image with a smaller quantisation step size. Its applications for adaptive source/channel coding and for coding fast changing sequences are examined.

APA, Harvard, Vancouver, ISO, and other styles

37

Kanu, Joseph. "Low bit-rate speech encoding for digital mobile radio." Thesis, Aston University, 1991. http://publications.aston.ac.uk/8089/.

Full text

Abstract:

The need for low bit-rate speech coding is the result of growing demand on the available radio bandwidth for mobile communications both for military purposes and for the public sector. To meet this growing demand it is required that the available bandwidth be utilized in the most economic way to accommodate more services. Two low bit-rate speech coders have been built and tested in this project. The two coders combine predictive coding with delta modulation, a property which enables them to achieve simultaneously the low bit-rate and good speech quality requirements. To enhance their efficiency, the predictor coefficients and the quantizer step size are updated periodically in each coder. This enables the coders to keep up with changes in the characteristics of the speech signal with time and with changes in the dynamic range of the speech waveform. However, the two coders differ in the method of updating their predictor coefficients. One updates the coefficients once every one hundred sampling periods and extracts the coefficients from input speech samples. This is known in this project as the Forward Adaptive Coder. Since the coefficients are extracted from input speech samples, these must be transmitted to the receiver to reconstruct the transmitted speech sample, thus adding to the transmission bit rate. The other updates its coefficients every sampling period, based on information of output data. This coder is known as the Backward Adaptive Coder. Results of subjective tests showed both coders to be reasonably robust to quantization noise. Both were graded quite good, with the Forward Adaptive performing slightly better, but with a slightly higher transmission bit rate for the same speech quality, than its Backward counterpart. The coders yielded acceptable speech quality of 9.6kbps for the Forward Adaptive and 8kbps for the Backward Adaptive.

APA, Harvard, Vancouver, ISO, and other styles

38

Haris, Muhammad. "Advanced modulation formats for high-bit-rate optical networks." Diss., Atlanta, Ga. : Georgia Institute of Technology, 2008. http://hdl.handle.net/1853/24811.

Full text

Abstract:

Thesis (Ph.D.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2008.
Committee Chair: Chang, Gee-Kung; Committee Co-Chair: Yu, Jianjun; Committee Member: Altunbasak, Yucel; Committee Member: Ji, Chunayi; Committee Member: Ralph, Stephen; Committee Member: Xu, Jun.

APA, Harvard, Vancouver, ISO, and other styles

39

Eryurtlu, Mehmet Omer Faruk. "New approaches in low bit rate image and video coding." Thesis, University of Surrey, 1995. http://epubs.surrey.ac.uk/842814/.

Full text

Abstract:

The aim of this thesis is to develop new image and video coding algorithms which can outperform the existing block transform-based standards at low bit rates. Instead of deep research in a particular approach, several different techniques are considered in order to be able to forecast the future in image and video compression. First, various block transform methods are examined, and their strengths and weaknesses are assessed. Then, classical methods such as linear prediction and vector quantisation are considered. Different linear prediction models and memoryless vector quantisation types are compared. The analysis-by-synthesis approach which has been successfully applied in speech coding, is used to design a linear predictive gain-shape vector quantiser. Since the main weakness of the block transform methods in low bit rates is the blocking effect which is visible at block boundaries, subband approach is considered as an alternative. Several aspects of filter design, region-of-support extension and decomposition structure are examined. Wavelet approach is also considered, however its application in digital image compression is not very different than the subband decomposition. A novel subband image and video codec which exploits the edge orientations in the lowest band in the adaptive vector quantisation of the higher bands is described. Moreover, another novel video codec based on predictive entropy coding of the gain-shape vector quantisation parameters is proposed. More importance is given to the promising segmentation-based techniques. Segmentation-based image coding is explained and several new techniques for contour representation, contour smoothing, edge profile smoothing and jagged edge rectification which improve the coding performance are applied. Then, a novel video coding algorithm based on joint region and motion segmentation is described. A number of control points, the locations of which can be predicted in video signals, are used to represent the region contours and another novel segmentation-based video codec using contour and texture prediction and working at very low bit rates is presented. Finally, the future of image and video coding is discussed and several research directions are recommended.

APA, Harvard, Vancouver, ISO, and other styles

40

Ong, Leh Kui. "Source reliant error control for low bit rate speech communications." Thesis, University of Surrey, 1994. http://epubs.surrey.ac.uk/843456/.

Full text

Abstract:

Contemporary and future speech telecommunication systems now utilise low bit rate (LBR) speech coding techniques in efforts to eliminate bandwidth expansion as a disadvantage of digital coding and transmission. These speech coders employ model-based approaches in compressing human speech into a number of parameters, using a well-known process known as linear predictive coding (LPC). However, a major side-effect observed in these coders is that errors in the model parameters have noticeable and undesirable consequences on the synthesised speech quality, and unless they are protected from such corruptions, the level of service quality will deteriorate rapidly. Traditionally, forward error correction (FEC) coding is used to remove these errors, but these require substantial redundancy. Therefore, a different perspective of the error control problems and solutions is necessary. In this thesis, emphasis is constantly placed on exploiting the constraints and residual redundancies present in the model parameters. It is also shown that with such source criteria in the LBR speech coders, varying degrees of error protection from channel corruptions are feasible. From these observations, error control requirements and methodologies, using both block- and parameter-orientated aspects, are analysed, devised and implemented. It is evident, that under the unusual circumstances which LBR speech coders have to operate in, the importance and significance of source reliant error control will continue to attract research and commercial interests. The work detailed in this thesis is focused on two LPC-based speech coders. One of the ideas developed for these two coders is an advanced zero redundancy scheme for the LPC parameters which is designed to operate at high channel error rates. Another concept proposed here is the use of source criteria to enhance the decoding capabilities of FEC codes to exceed that of maximum likelihood decoding performance. Lastly, for practical operation of LBR speech coders, lost frame recovery strategies are viewed to be an indispensable part of error control. This topic is scrutinised in this thesis by investigating the behaviour of a specific speech coder under irrecoverable error conditions. In all of the ideas pursued above, the effectiveness of the algorithms formulated here are quantified using both objective and subjective tests. Consequently, the capabilities of the techniques devised in this thesis can be demonstrated, examples of which are: (1) higher speech quality produced under noisy channels, using an improved zero-redundancy algorithm for the LPC filter coefficients; (2) as much as 50% improvement in the residual BER and decoding failures of FEC schemes, through the utilisation of source criteria in LBR speech coders; and (3) acceptable speech quality produced under high frame loss rates (14%), after formulating effective strategies for recovery of speech coder parameters. It is hoped that the material described here provide concepts which can help achieve the ideals of maximum efficiency and quality in LBR speech telecommunications.

APA, Harvard, Vancouver, ISO, and other styles

41

Koestoer, Nanda Prasetiyo, and npkoestoer@yahoo com au. "Robust Linear Prediction Analysis for Low Bit-Rate Speech Coding." Griffith University. School of Microelectronic Engineering, 2002. http://www4.gu.edu.au:8080/adt-root/public/adt-QGU20030407.142552.

Full text

Abstract:

Speech coding is a very important area of research in digital signal processing. It is a fundamental element of digital communications and has progressed at a fast pace in parallel to the increase of demands in telecommunication services and capabilities. Most of the speech coders reported in the literature are based on linear prediction (LP) analysis. Code Excited Linear Predictive (CELP) coder is a typical and popular example of this class of coders. This coder performs LP analysis of speech for extracting LP coefficients and employs an analysis-by-synthesis procedure to search a stochastic codebook to compute the excitation signal. The method used for performing LP analysis plays an important role in the design of a CELP coder. The autocorrelation method is conventionally used for LP analysis. Though this works reasonably well for noise-free (clean) speech, its performance goes down when signal is corrupted by noise. Spectral analysis of speech signals in noisy environments is an aspect of speech coding that deserves more attention. This dissertation studies the application of recently proposed robust LP analysis methods for estimating the power spectrum envelope of speech signals. These methods are the moving average, moving maximum and average threshold methods. The proposed methods will be compared to the more commonly used methods of LP analysis, such as the conventional autocorrelation method and the Spectral Envelope Estimation Vocoder (SEEVOC) method. The Linear Predictive Coding (LPC) spectrum calculated from these proposed methods are shown to be more robust. These methods work as well as the conventional methods when the speech signal is clean or has high signal-to-noise ratio. Also, these robust methods give less quantisation distortion than the conventional methods. The application of these robust methods for speech compression using the CELP coder provides better speech quality when compared to the conventional LP analysis methods.

APA, Harvard, Vancouver, ISO, and other styles

42

Huang, Weiyuan. "ESTELLE verification of ATM available bit rate (ABR) control protocol." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2001. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/MQ59326.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

43

May, Richard John. "Perceptual content loss in bit rate constrained IFS encoded speech." Thesis, University of Portsmouth, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.396323.

Full text

APA, Harvard, Vancouver, ISO, and other styles

44

Samingan, Ahmad Kamsani. "Minimum bit error rate multiuser detection techniques for DS-CDMA." Thesis, University of Southampton, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.398594.

Full text

APA, Harvard, Vancouver, ISO, and other styles

45

Salami, Redwan Ali. "Robust low bit rate analysis-by-synthesis predictive speech coding." Thesis, University of Southampton, 1990. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.277700.

Full text

APA, Harvard, Vancouver, ISO, and other styles

46

Mizuki, Marcelo Mikiyo. "Edge based video image compression for low bit rate applications." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/10645.

Full text

Abstract:

Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.
Includes bibliographical references (p. 94-95).
by Marcelo Mikiyo Mizuki.
M.S.

APA, Harvard, Vancouver, ISO, and other styles

47

Liang, Yang, and 梁洋. "Frame Rate Control for Constant Bit Rate Video." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/33317929075882486255.

Full text

Abstract:

碩士
國立臺北科技大學
電機工程系碩士班
91
This thesis proposed a variable frame skip (VFS) mechanism to improve the video quality in low bit rate channel. The basic idea of VFS mechanism is to decide and skip a suitable, non-fixed number of frames in temporal domain to reduce bit usage. The saved bits can be allocated to enhance the spatial quality of the video. In literature, several methods of frame skip decision have proposed, but most of them only consider the similarity between neighboring coded frames as the decision criterion. Our proposed method takes into account the reconstruction of the skipped frames using motion-compensated frame interpolation at decoder. The proposed VFS models the reconstructed quality of the skipped frame and, therefore, can provide a fast estimate to the frame skipped at encoder. The proposed VFS can decide the frame skip in real time, and its encoded video has better spatial-temporal bit allocation.

APA, Harvard, Vancouver, ISO, and other styles

48

Yen, Lee-Lung, and 顏立隆. "VARIABLE BIT RATE IMBE VOCODER." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/ajscxk.

Full text

Abstract:

碩士
大同大學
通訊工程研究所
102
The improved multi-band excitation (IMBE) vocoder is developed from the multi-band excitation (MBE) speech model. It can provide good speech quality with low bit rate, primarily due to its special form of treating excitation signals. Specifically, for each speech frame, through spectral analysis, the IMBE vocoder divides the whole spectrum into sub-bands, each consisting three harmonics. Then each sub-band is discriminated as voiced or unvoiced. By doing this, the IMBE vocoder allows a speech frame having voiced and unvoiced sound at the same time, i.e., the nature of mixed sound. And this is why the IMBE vocoder can reconstruct synthetic speech with more nature quality. In this thesis, for the IMBE vocoder, with decreasing upper limited number of sub-bands (the original setting is 12), we investigated the corresponding bit-rate reduction and the possible quality degradation of synthesized speech. Our experiments proved that, with the 4000 Hz system bandwidth, high-pitched female voice will be affected less due to the decreasing of sub-bands. For example, only 8 sub-bands are required for a voice with 170 Hz fundamental frequency. On the contrary, more than 12 sub-bands are required for voice with fundamental frequency lower than 121 Hz. Our experiments also showed that, within acceptable speech intelligibility, the bit-rate reduction can be achieved if a certain amount of speech quality degradation is tolerable.

APA, Harvard, Vancouver, ISO, and other styles

49

Chiang, Chun-Yi, and 江俊毅. "Variable Bit-Rate Color Image Quantization." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/16912230493175865794.

Full text

Abstract:

碩士
靜宜大學
資訊管理學系研究所
98
As the images between neighboring pixels high similarity, so based on the characteristics, we proposed two methods of variable bit-rate about color image quantization in this paper. The first method focuses on the degree of similarity between the top and left pixel for encoding pixel to determine the state of the palette to use size. Using this approach we hope to find a better palette color in a short time. The second method splits the given image into equilateral and fixed-size blocks and then calculates the degree of similarity between pixels in every split block. If the similar degree is high, this block only takes a common representative color in palette. On the contrary if the similar degree of block is widely different, the comparison that this block must do the similar intensity once again after splitting, until it is unable to split again. Our method not only can solve the conventional color image quantization that different bit-rate would correspond to a different size of palette, but also under conditions similar to the bit-rate the quality of reconstructed images are better than traditional methods.

APA, Harvard, Vancouver, ISO, and other styles

50

Chatterjee, Shoma. "Low bit-rate facial video compression." Thesis, 2001. http://localhost:8080/iit/handle/2074/3296.

Full text

APA, Harvard, Vancouver, ISO, and other styles

Dissertations / Theses on the topic 'Bit rate'

Create a spot-on reference in APA, MLA, Chicago, Harvard, and other styles