This paper proposes a new model-based method for transform coding of audio signals. The input signal is mapped in "perceptual" domain by linear-predictive weighting filter followed by modified discrete cosin...
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This paper proposes a new model-based method for transform coding of audio signals. The input signal is mapped in "perceptual" domain by linear-predictive weighting filter followed by modified discrete cosine transform (MDCT). To provide bitstream scalability, model-based bit plane coding is then applied with respect to the mean square error (MSE) criterion. We present methods to estimate the symbol probability in bit planes assuming a generalized Gaussian model for the distribution of MDCT coefficients. We compare the performance of the proposed bitstream scalable coder with stack-run coding and ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is equivalent to or slightly worse than reference coders, but presents the nice advantage of being scalable. Performance penalty due to bitstream scalability is evident at low bitrates.
There are many high performance speech coding algorithms, but most of them using bit rates higher than 5kbps. Expected quality of low bit rate speech coding has not been achieved yet, which is an area of interest for ...
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ISBN:
(纸本)9781629938219
There are many high performance speech coding algorithms, but most of them using bit rates higher than 5kbps. Expected quality of low bit rate speech coding has not been achieved yet, which is an area of interest for military and security applications. There is still room for improvement for low bit rates. Low bit rate speech coding using sinusoidal transform coding is discussed in this paper. The sinusoidal transform coding (STC) is a frequency domain speech compression technique, which can produce speech of high quality at bit rates below *** STC, speech signal are represented by linear combination of sine waves with time-vary amplitudes, phase, and frequencies. STC can produce high quality synthesized speech at low bit rates. This paper focuses on the development of a fully speech coder at 2.4 kbps and 4.8 kbps.
The discrete Fourier transform (DFT) has been proposed as a suitable technique for data transmission over channels that introduce impulse noise. However, in the presence of band-limited additive white Gaussian noise (...
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The discrete Fourier transform (DFT) has been proposed as a suitable technique for data transmission over channels that introduce impulse noise. However, in the presence of band-limited additive white Gaussian noise (AWGN), the performance of DFT coding is degraded. This paper deals with techniques for single impulse correction and combating the detrimental effect of band-limited AWGN.
Demand for screen content videos that contain computer generated text and graphics is growing. They are very different from natural videos, because they include much sharper edge transitions and very repetitive patter...
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ISBN:
(纸本)9781467399623
Demand for screen content videos that contain computer generated text and graphics is growing. They are very different from natural videos, because they include much sharper edge transitions and very repetitive patterns. On this type of material, the efficacy of the conventional discrete cosine transform (DCT) is questionable because it relies on the assumption that a Gauss-Markov model leads to a base-band signal. However, the assumption may not hold true for screen content material. This work exploits a class of staircase transforms. Unlike the DCT whose bases are samplings of sinusoidal functions, the staircase transforms have their bases sampled from staircase functions, which better approximate the sharp transitions often encountered in the context of screen content. The staircase transform is integrated into a hybrid transform coding scheme, in conjunction with DCT. It is experimentally shown that the proposed approach provides an average of 2.9% compression performance gains in terms of BD-rate reduction. A perceptual comparison further demonstrates that the use of staircase transform achieves substantial reduction in ringing artifact due to the Gibbs phenomenon.
This paper proposes a hybrid transform coding technique, where the intraframes of video sequence are coded by discrete wavelet transform and the interframes are coded with discrete cosine transform technique. It also ...
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This paper proposes a hybrid transform coding technique, where the intraframes of video sequence are coded by discrete wavelet transform and the interframes are coded with discrete cosine transform technique. It also proposes the selection of sequence of frames predicted from the reference intraframes. This proposal consistently minimizes the prediction error for the predicted frame for further processing. The experimental results show that the proposed hybrid transformation coding technique outperforms over conventional transformation coding technique in terms of encoding time and prediction errors.
The main advantage of backward over forward adaptive coding schemes is to update the coding parameters with the data available at the decoder, avoiding thereby any excess bit rate. The performances of two practical ba...
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The main advantage of backward over forward adaptive coding schemes is to update the coding parameters with the data available at the decoder, avoiding thereby any excess bit rate. The performances of two practical backward adaptive transform coding schemes are analyzed in terms of rate and distortion for two transforms: the KLT (Karhunen-Loeve transform) and the LDU transform (based on a lower-diagonal-upper factorization of the covariance matrix, R, of the data). For both algorithms, we model the expected distortion w.r.t. the number of vectors available at the decoder. Our analysis shows that, for an algorithm using Sheppard's correction on the second order moment estimates, the distortion should converge to the target distortion. Without this correction, the effects of backward adaptation are shown to move the actual r(D) point of the system from the target point by the same term for both transforms. Simulation results confirming the theoretic analysis are presented.
In MPEG, the image sequence compression is achieved by motion compensation, transformation, quantization and entropy coding. In this paper, we follow the same path by using wavelet based modules in the coding chain. T...
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ISBN:
(纸本)0780333365
In MPEG, the image sequence compression is achieved by motion compensation, transformation, quantization and entropy coding. In this paper, we follow the same path by using wavelet based modules in the coding chain. The prediction error signal is obtained by using variable block size motion compensation (VBSMC). Great attention has been paid to smooth the motion vector field, which helps to increase the coding efficiency. After VBSMC, discrete wavelet transform (DWT) is then applied to the prediction error signal to obtain wavelet coefficients. Some new schemes are introduced. First, a new and effective hybrid coder using quadtree and lattice vector quantization (QTLVQ) is introduced, which can efficiently code the wavelet coefficients of the motion compensated prediction error signal. Second, a new rate allocation scheme different from traditional ones is introduced, which can flexibly control the coder's output rate and the duality of reconstructed video signal.
A new, oddly stacked, critically sampled, single side-band (SSB) [7] analysis/synthesis system based on Time Domain Aliasing Cancellation (TDAC) [1],[2] is described in this paper. The specifications for the analysis ...
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A new, oddly stacked, critically sampled, single side-band (SSB) [7] analysis/synthesis system based on Time Domain Aliasing Cancellation (TDAC) [1],[2] is described in this paper. The specifications for the analysis and synthesis filter responses are developed and a number of designs which satisfy the reconstruction requirements are described. The application of TDAC systems to Subband/transform coding is also discussed and the objective performance of a 32 band coder using several different window designs is presented and compared with a coder based on Frequency Domain Aliasing Cancellation (FDAC) filter banks [3]-[5].
In using partial transform for the coding of hyperspectral image, it must consider spectral decorrelation between image components, issues that a combined adaptive classification and partial transform algorithm for hy...
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In using partial transform for the coding of hyperspectral image, it must consider spectral decorrelation between image components, issues that a combined adaptive classification and partial transform algorithm for hyperspectral image compression is presented in this paper. Our method uses a linear prediction based on adaptive classification to decorrelate the spectrum redundancy and a 2D integer reversible DCT-based scheme as spatial compression engine. The classification includes band ordering, band regrouping and reference frame selection. Experimental results on AVIRIS data indicate that the proposed approach is a novel low complexity lossy hyperspectral image compression scheme and exhibits performance better than other comparative methods in quality and fidelity. It can be used for hyperspectral image compression in the embedded processor of the platform on satellite or aircraft
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