A challenging and important research problem in speech coding emerged recently with the need for a 16 kb/s speech coding algorithm that has very low coding delay while achieving essentially the same high quality as th...
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A challenging and important research problem in speech coding emerged recently with the need for a 16 kb/s speech coding algorithm that has very low coding delay while achieving essentially the same high quality as the 32 kb/s ADPCM standard G.721. Although speech coding algorithms developed in the last few years are able to provide the required quality at 16 kb/s, these coders introduce a substantial delay, in most cases due to forward adaptation where input speech samples are buffered to compute speech modeling parameters prior to actual coding of the samples. To meet the low-delay requirement, forward adaptation is not feasible, yet backward adaptation at low rates tends to cause degraded quality and severe propagation of bit errors. Although the well-established ADPCM algorithm satisfied the delay and quality objectives at 32 kb/s, its quality at 16 kb/s is unacceptable. In this paper, we describe low-delay vector excitation coding (LD-VXC), a new coding algorithm which provides high quality with less than 2 ms of coding delay and is robust to transmission errors. The algorithm combines techniques such as vector quantization, analysis-by-synthesis, perceptual weighting, together with backward adaptive linear predictive encoding, and uses a novel long-term predictor employing backward adaptive pitch tracking. Perceptually based nose shaping and postfiltering contribute to the masking of audible quantization noise.
The penalty incurred by imposing a finite delay constraint in lossless source coding of a memoryless source is investigated. It is well known that for the so-called block-to-variable and variable-to-variable codes, th...
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The penalty incurred by imposing a finite delay constraint in lossless source coding of a memoryless source is investigated. It is well known that for the so-called block-to-variable and variable-to-variable codes, the redundancy decays at best polynomially with the delay, where in this case the delay is identified with the source block length or maximal source phrase length, respectively. In stark contrast, it is shown that for sequential codes (e. g., a delay-limited arithmetic code) the redundancy can be made to decay exponentially with the delay constraint. The corresponding redundancy-delay exponent is shown to be at least as good as the Renyi entropy of order 2 of the source, but (for almost all sources) not better than a quantity depending on the minimal source symbol probability and the alphabet size.
The authors show that for linear block codes defined over extensions of GF(2) a variant of the sub-optimal soft-decision Dorsch algorithm (1974) offers very good performance with low complexity. Furthermore, we show t...
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The authors show that for linear block codes defined over extensions of GF(2) a variant of the sub-optimal soft-decision Dorsch algorithm (1974) offers very good performance with low complexity. Furthermore, we show that, owing to the nature of the algorithm, it can easily be adapted to produce soft-decision output. This is then exploited in an iterative decoding scheme for product codes which is based on a decoding algorithm first proposed by Pyndiah et nl (1996, 1998), Two different interleaver structures are presented yielding different performances both in terms of coding delay and bit-error-rate for a given signal-to-noise ratio. Reed-Solomon codes are used in the simulations. For reasons of complexity, only codes defined over GF(16) are considered. Simulations were carried out for the AWGN channel.
This paper presents three new global positioning system acquisition methods that make use of both L1 C/A and L2C signals in a combined way. The methods perform joint estimation of the Doppler frequencies and code dela...
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This paper presents three new global positioning system acquisition methods that make use of both L1 C/A and L2C signals in a combined way. The methods perform joint estimation of the Doppler frequencies and code delays on L1 and L2 without increasing the coherent integration time. Each method is assessed in terms of theoretical probabilities of false alarm and detection as well as through testing with real intermediate frequency data. The first method is a noncoherent summation of L1 and L2 correlator outputs. The second method implements an independent differential summation of L1/L2 correlator outputs, and the third method uses a noncoherent plus dependent differential summation. While each method provides increased detection performance compared with the standard noncoherent acquisition applied on L1 C/A, the noncoherent plus dependent differential summation method outperforms all of the others in the scenarios investigated.
Recently subband coding of images has received considerable attention and has become an effective tool for image compression. In recent work by Karlsson and Vetterli, a 3-D subband coding of video using 2-tap symmetri...
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Recently subband coding of images has received considerable attention and has become an effective tool for image compression. In recent work by Karlsson and Vetterli, a 3-D subband coding of video using 2-tap symmetrical short-kernel filters for vertical filtering was suggested for transmission over packet-switched network environments. In this paper, a modified short-kernel Alter pair is proposed to eliminate the coding delay and to improve the reconstructed image quality. Simulations are conducted and satisfactory results are achieved.
The source-channel coding problem for point-to-point communications is reviewed. This is first done with a critical eye on the separation principle. It is pointed out that when we design a system according to the sepa...
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The source-channel coding problem for point-to-point communications is reviewed. This is first done with a critical eye on the separation principle. It is pointed out that when we design a system according to the separation principle we end up with and end-to-end map which is essentially deterministic. It is indeed the fact that this map is deterministic what makes it inevitable that the separation principle leads to long codes and thus large delays. It is pointed out that in some applications we don't need or even don't want deterministic mappings. When this is the case one can go beyond the separation principle and aim at systems that have smaller (or eve zero) delay. This is obtained by matching the source and the channel in a probabilistic way. A speculative example suggested by biological communications is given.
IEEE 802.11s-based infrastructure Wireless Mesh Networks (iWMNs) are envisaged as a promising solution to provide ubiquitous wireless Internet access. The limited network capacity is a problem mainly caused by the med...
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IEEE 802.11s-based infrastructure Wireless Mesh Networks (iWMNs) are envisaged as a promising solution to provide ubiquitous wireless Internet access. The limited network capacity is a problem mainly caused by the medium contention between mesh users and the mesh access points (MAPs), which gets worst when the mesh clients employ the Transmission Control Protocol (TCP). To mitigate this problem, we use wireless network coding (WNC) in the MAPs. The aim of this proposal is to take advantage of the network topology around the MAPs, to alleviate the contention and maximize the use of the network capacity. We evaluate WNC when is used in MAPs. We model the formation of coding opportunities and, using computer simulations, we evaluate the formation of such coding opportunities. The results show that as the users density grows, the coding opportunities increase up to 70%;however, at the same time, the coding delay increments significantly. In order to reduce such delay, we propose to adaptively adjust the time that a packet can wait to catch a coding opportunity in an MAP. We assess the performance of moving-average estimation methods to forecast this adaptive sojourn time. We show that using moving-average estimation methods can significantly decrease the coding delay since they consider the traffic density conditions.
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