The authors developed a new adaptive image coding technique using interpolative mixed vector quantization (VQ) which mixes the 2D discrete cosine transform (DCT) with VQ. A median filter is first used to eliminate fin...
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The authors developed a new adaptive image coding technique using interpolative mixed vector quantization (VQ) which mixes the 2D discrete cosine transform (DCT) with VQ. A median filter is first used to eliminate fine noise while not smearing the edges. After subsampling, the image is encoded by an adaptive mixed VQ coding scheme. At the receiver, a finite-impulse-response (FIR) median hybrid filter is used to interpolate missing pixels after subsampling. Some simulation results are given. It was found that high-quality coded images were obtained at a bit rate from about 0.25 to 0.5 b.p.p.< >
The authors propose an efficient vector quantization scheme and a novel linear predictive coding (LPC) analysis scheme, both of which exploit interframe correlation in the successive spectrum envelope of speech signal...
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The authors propose an efficient vector quantization scheme and a novel linear predictive coding (LPC) analysis scheme, both of which exploit interframe correlation in the successive spectrum envelope of speech signals. The first quantization scheme proposed is a multistage vector quantization of line spectrum pair (LSP) parameters with a partially adaptive codebook (MSVQ-AC). The second algorithm is an LPC analysis scheme, with closed-loop adaptive prefiltering (LPC-PF), which realized temporary higher-order analysis than the standard LPC with a few additional transmission bits. A combined system of the LPC-PF and two-split, two-stage VQ with the adaptive codebook can quantize tenth-order LSP parameters at around 23 bits/frame, realizing sufficient quality and reasonable complexity.< >
The discrete neural network model due to Hopfield (1982), and his following developments, allow to tie the dynamical evolution of a neural network to a quantity-the energy of the network-which is monotonically decreas...
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The discrete neural network model due to Hopfield (1982), and his following developments, allow to tie the dynamical evolution of a neural network to a quantity-the energy of the network-which is monotonically decreasing up to a minimum stable point. A network of this kind is able to ''solve'' efficiently some useful computational problems. An application of the neural Hopfield model to the source (optimal) coding problem is shown;this application gives the structure of a Hopfield network, which is able to calculate the lengths of the codewords that must be associated to a source alphabet so as to minimize their average length. The case of a stationary and memoryless block-to-variable-length (B-VL) source coding is discussed.
This paper presents an adaptive coding scheme for digital communication over time-varying channels. The scheme is based on a finite-state Markov channel model. Emphasis is on the adaptation of the error protection to ...
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This paper presents an adaptive coding scheme for digital communication over time-varying channels. The scheme is based on a finite-state Markov channel model. Emphasis is on the adaptation of the error protection to the actual channel state. The throughput gains that are achieved by the adaptive scheme relative to the conventional nonadaptive coding methods are demonstrated by several examples. Of special interest is the use of punctured convolutional codes with maximum-likelihood Viterbi algorithm to enable adaptive encoding and decoding without modifying the basic structure of the encoder and the decoder.
The design of an efficient mobile data link protocol (MDLP) based on a Type II hybrid automatic request for repetition (ARQ) scheme using convolutional coding and code combining is presented. An adaptive coding scheme...
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The design of an efficient mobile data link protocol (MDLP) based on a Type II hybrid automatic request for repetition (ARQ) scheme using convolutional coding and code combining is presented. An adaptive coding scheme for the header is proposed and analyzed. Simulation results show significant improvement in throughput at low SNR as compared to conventional schemes. Precise specification of the MDLP using a formal description technique based on Estelle is outlined and discussed.
Type I hybrid ARQ systems use an error-control code of fixed rate for error-correction in each transmitted packet. In such systems, the code rate must be carefully chosen to match the channel bit error rate (BER) in o...
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Type I hybrid ARQ systems use an error-control code of fixed rate for error-correction in each transmitted packet. In such systems, the code rate must be carefully chosen to match the channel bit error rate (BER) in order to maximize system throughput. Whenever there is a mismatch between code rate and channel BER, the throughput suffers. This paper proposes a type I hybrid ARQ system which automatically adjusts its code rates to match channel BERs. The proposed system is very simple to implement and yet performs significantly better than type I hybrid ARQ systems of fixed code rate.
The design of millimeter-wave filtering devices using dielectric resonators of various shapes in nonradiative dielectric (NRD) waveguides is considered. By developing different approximate and rigorous theoretical met...
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The design of millimeter-wave filtering devices using dielectric resonators of various shapes in nonradiative dielectric (NRD) waveguides is considered. By developing different approximate and rigorous theoretical methods, the resonant and coupling properties (frequencies, field configuration, unloaded and loaded quality factors, scattering parameters, etc.) are modeled and implemented in an adaptive code. Experimental comparison through suitable measurements are performed to validate the theoretical and numerical results.< >
This paper describes the construction of a universal code for minimizing L.D. Davisson's minimax redundancy in a range where the true model and stochastic parameters are unknown.
This paper describes the construction of a universal code for minimizing L.D. Davisson's minimax redundancy in a range where the true model and stochastic parameters are unknown.
作者:
GALAND, CRMENEZ, JEROSSO, MMUNIV NICE
SOPHIA ANTIPOLISFRANCE LASSY
CNRSURA 1376EQUIPE 135F-06041 NICEFRANCE IBM CORP
THOMAS J WATSON RES CTRDEPT COMP SCIYORKTOWN HTSNY 10598 IBM CORP
DEPT ARCHITECTURE & TELECOMMUNRES TRIANGLE PKNC
Since its recent introduction by Atal and Schroeder, the code excited linear predictive (CELP) coder has been thoroughly and widely studied by the speech coding research community, and has already been adapted to seve...
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Since its recent introduction by Atal and Schroeder, the code excited linear predictive (CELP) coder has been thoroughly and widely studied by the speech coding research community, and has already been adapted to several standards for telephone speech coding. The CELP algorithm represents a breakthrough in speech coding, for it encodes telephone quality speech at 8 kb/s without noticeable distortion. Previously, this performance was achievable by coders operating at 16 kb/s or higher. However, the drawback of the CELP is its inherent complexity, which, despite the fast progress of the technology, may represent a problem of cost or feasibility in products. In this paper, we discuss a new way to consider the CELP concept, which allows us to cut the processing load while keeping the same speech quality. Rather than performing the individual weighting of each candidate sequence, we propose a global implementation of the perceptual weighting function at the codebook level. As a result, the analysis-by-synthesis procedure does not require the processing of all the candidate sequences through the synthesis and weighting filters, and therefore the complexity requirement of the algorithm is much reduced. The new concept is carried out with an adaptive codebook. We report on two fixed-point implementations of our adaptive CELP (ACELP) algorithm: a 7.2 kb/s block coder (7 MIPS), and a 12 kb/s low-delay coder (11 MIPS). Both coders have been rated to provide the same quality as the 13 kb/s block coder adopted by the GSM for the European cellular telephone.
The discrete cosine-III transform (DC3T) is the same as the discrete symmetric cosine transform (DSCT) with a specific preprocessing of input data. It has less computational complexity than the discrete cosine transfo...
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The discrete cosine-III transform (DC3T) is the same as the discrete symmetric cosine transform (DSCT) with a specific preprocessing of input data. It has less computational complexity than the discrete cosine transform (DCT) in terms of multiplications. The DC3T is also related to the DCT by a weighting matrix. The performance of the DC3T is compared to the DCT when compression is performed by adaptive coding. Experimental results show that a significant improvement in visual performance and mean square reconstruction error can be achieved over what is possible with the DCT. The better performance is attributed to the weighting of the DCT coefficients, which is indirectly achieved at reduced computational cost.
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