This paper introduces concept of G.723.1 speech coder and analyses its technology and features. We advise to optimize its running time of G.723.1 speech coder. We put forward to improve some modules with large computa...
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This paper introduces concept of G.723.1 speech coder and analyses its technology and features. We advise to optimize its running time of G.723.1 speech coder. We put forward to improve some modules with large computational complexity, such as pitch estimation module, the adaptive and the fixed codebook research modules.
This paper presents a set of descriptors for On-line signature writer identification. These descriptors are intended to be used in e-business and e-government to detect signature forgery where it is hard to identify t...
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Geveze software is one of many implementations in text-to-speech synthesis for various languages. The program is based on vocal tract modeling and compresses speech by the LPC method. During synthesis, for each letter...
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Geveze software is one of many implementations in text-to-speech synthesis for various languages. The program is based on vocal tract modeling and compresses speech by the LPC method. During synthesis, for each letter of a given word, the nearest combination of the letter sequences within the words used in training is searched and its parameters are used. As in other systems based on vocal tract modeling, a pulse train generates excitation for voiced sounds, while a noise signal is used for unvoiced sounds. The obtained signal is then amplified with a coefficient special to the sound at that instant and finally sent to an IIR filter, whose filter characteristics are determined by LPC coefficients, and the digitized waveform of the speech is obtained. During training, 10 LPC coefficients, 1 gain, and 1 period information bit are obtained for each 25 ms window, separated by 10 ms. During synthesis, these values change every 10 ms to the values of the following window. The digital signal at the output of the IIR filter is converted to analog, which has to be passed through a low pass filter (LPF) in order to smooth the transitions between windows. After filtering, the analog signal is ready to be amplified. Our objective is to design this system, already running on computer, as an integrated circuit and, if possible, to have a single chip solution with optimum cost and performance.
The objective of this work is to investigate whether joint optimization of short-term and long-term predictors manifests significant advantages over the sequential optimization in speech coding. We propose a new joint...
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The objective of this work is to investigate whether joint optimization of short-term and long-term predictors manifests significant advantages over the sequential optimization in speech coding. We propose a new joint optimization method based on Wiener filtering. The proposed analysis model resolves the pitch-bias problem of classical LPC analysis by considering the contribution of the long-term predictor while optimizing the short-term predictor. Our approach to joint optimization is based on analysis-by-synthesis and guarantees the synthesis filter stability. By applying our proposed joint optimization approach to CELP coding we obtain superior objective and subjective performance relative to CELP coding with sequential optimization. To provide voice quality equivalent to that of sequentially optimized CELP, the jointly optimized coder needs fewer FCB pulses and requires a reduced bit budget for LPC quantization. Our listening tests suggest that the JCELP coder at 4.25 kbps is equivalent in quality to the G.729 at 8 kbps.
Discover the basic telecommunications systems principles in an accessible learn-by-doing format Communication Systems Principles Using MATLAB covers a variety of systems principles in telecommunications in an accessib...
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ISBN:
(数字)9781119470687
ISBN:
(纸本)9781119470670
Discover the basic telecommunications systems principles in an accessible learn-by-doing format Communication Systems Principles Using MATLAB covers a variety of systems principles in telecommunications in an accessible format without the need to master a large body of theory. The text puts the focus on topics such as radio and wireless modulation, reception and transmission, wired networks and fiber optic communications. The book also explores packet networks and TCP/IP as well as digital source and channel coding, and the fundamentals of data encryption. Since MATLAB® is widely used by telecommunications engineers, it was chosen as the vehicle to demonstrate many of the basic ideas, with code examples presented in every chapter. The text addresses digital communications with coverage of packet-switched networks. Many fundamental concepts such as routing via shortest-path are introduced with simple and concrete examples. The treatment of advanced telecommunications topics extends to OFDM for wireless modulation, and public-key exchange algorithms for data encryption. Throughout the book, the author puts the emphasis on understanding rather than memorization. The text also: Includes many useful take-home skills that can be honed while studying each aspect of telecommunications Offers a coding and experimentation approach with many real-world examples provided Gives information on the underlying theory in order to better understand conceptual developments Suggests a valuable learn-by-doing approach to the topic Written for students of telecommunications engineering, Communication Systems Principles Using MATLAB® is the hands-on resource for mastering the basic concepts of telecommunications in a learn-by-doing format.
This paper introduces a new computational algorithm for the partial correlation coefficients of a linear system given the covariance of its output when excited by a white input noise. Although derived from Levinson...
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This paper introduces a new computational algorithm for the partial correlation coefficients of a linear system given the covariance of its output when excited by a white input noise. Although derived from Levinson's well-known procedure, the proposed algorithm does not make use of the usual parameters in the linear prediction recursion. It may be implemented using fixed point arithmetics. Application to speech waves is emphasized.
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