Error correction codes are used widely in all wireless communication systems to reduce data corruption. The most widely used decoding algorithm is the Viterbi decoder which is used with different parameters for differ...
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ISBN:
(纸本)9781424410934
Error correction codes are used widely in all wireless communication systems to reduce data corruption. The most widely used decoding algorithm is the Viterbi decoder which is used with different parameters for different standards requirements. This paper analyses the different Viterbi decoders and implements a reconfigurable adaptive Viterbi decoder for GPRS, EDGE and Wimax technologies. The high performance genetic soft input hard output Viterbi decoder is prototyped on a FPGA.
A novel and efficient speckle noise reduction algorithm based on Bayesian contourlet shrinkage using contourlet transform is ***,we show the sub-band decompositions of SAR images using contourle transforms,which provi...
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A novel and efficient speckle noise reduction algorithm based on Bayesian contourlet shrinkage using contourlet transform is ***,we show the sub-band decompositions of SAR images using contourle transforms,which provides sparse representation at both spatial and directional ***,a Bayesian contourlet shrinkage factor is applied to the decomposed data to estimate the best value for noise-free contourle *** results show that compared with conventional wavelet despeckling algorithm,the proposed algorithm can achieve an excellent balance between suppresses speckle effectively and preserve image details,and the significant information of origina image like textures and contour details is well ma intained.
This paper presents a high-quality real-time pitch-shifting algorithm with a time-varying factor for monophonic audio and musical signals. The pitch-shifting algorithm is based on the resampling and time-scale modific...
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ISBN:
(纸本)9788890147913
This paper presents a high-quality real-time pitch-shifting algorithm with a time-varying factor for monophonic audio and musical signals. The pitch-shifting algorithm is based on the resampling and time-scale modification method. A new time-scale modification method has been developed which is called the Normalized Filtered Correlation Time-Scale Modification (NFC-TSM) method. It uses a ring buffer for time-scaling. The best splicing point is searched in the normalized low-pass filtered signal using the Average Magnitude Difference Function (AMDF). The new method results in low-latency and high-quality pitch-shifting of musical signals.
Sound synthesis by block‐based physical modeling of musical instruments separates the tasks of component modeling and managing their interactions. The components are the exciters or the resonators, and their interact...
Sound synthesis by block‐based physical modeling of musical instruments separates the tasks of component modeling and managing their interactions. The components are the exciters or the resonators, and their interactions are managed by explicit interaction blocks, which are obtained from the physical continuity and energy conservation rules. Well‐known examples of the interactors include the wave‐ digital adaptors and the digital waveguide scattering junctions. When the virtual instruments need to be interfaced to the outside environment with sensors and actuators for bidirectional interaction, it is advantageous to reformulate the interactors to accept and provide signal inputs and outputs, respectively. In this contribution, we refer to these elements as nodes, introduce different types of nodes, and discuss their interconnection.
We present an effective and simple noise robust front-end based on post-processing of standard mel-frequency cepstral coefficients (MFCCs) features. A feature processing method consisting of silence energy normalizati...
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ISBN:
(纸本)0980326702
We present an effective and simple noise robust front-end based on post-processing of standard mel-frequency cepstral coefficients (MFCCs) features. A feature processing method consisting of silence energy normalization, cepstral mean and variance normalization, and Auto-Regression and Moving-Average (ARMA) filtering in both log-energy and cepstral domain for noise-robust feature extraction is proposed. This technique is named SEN-MVN-A processing. From, the experiments conducted on Aurora 2.0 database, we showed that SEN-MVN-A provides an averaged improvement of word recognition accuracy of 15.7% , 2.1.5% and 6.4% for test sets A, B and C, respectively, when compared with the baseline results.
In statistical machine translation (SMT) research, phrase-based methods have been receiving more interest in recent years. In this paper, we first give a brief survey of phrase-based SMT framework, and then make detai...
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ISBN:
(纸本)9783540709381
In statistical machine translation (SMT) research, phrase-based methods have been receiving more interest in recent years. In this paper, we first give a brief survey of phrase-based SMT framework, and then make detailed comparisons of two typical implementations: alignment template approach and standard phrase-based approach. At last, we propose an improved model to integrate alignment template into standard phrase-based SMT as a new feature in a log-linear model. Experimental results show that our method outperforms the baseline method.
In this paper, the application of a well known mathematical theorem, Banach's fixed point theorem [1], is investigated in iterative signalprocessing in communications. In most practical communication systems some...
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Room reverberation consists of a multitude of reflections from surfaces and objects in a room. Particularly the late reverberation tail resembles noise with an exponential decay envelope. Artificial reverberation algo...
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Room reverberation consists of a multitude of reflections from surfaces and objects in a room. Particularly the late reverberation tail resembles noise with an exponential decay envelope. Artificial reverberation algorithms try to simulate this in a computationally efficient manner. Some proposed algorithms are based on the convolution with a sparse FIR filter corresponding to a randomized sparse sequence of unit impulses. In this paper we search for such sequences with minimal impulse density vs. maximal smoothness of the noise-like characteristics. Such noise is called here "velvet noise", because it can sound smoother than the Gaussian noise. The perceptual characteristics of velvet noise are described by results from listening experiments and auditory analysis. Reverb algorithms based on velvet noise are discussed and analyzed.
A novel interferometric inverse synthetic aperture radar (InISAR) imaging method, include an efficient phase unwrapping approach, is proposed, which can work very well under very low signal noise ratio condition. Base...
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The term "color temperature" represents the color of light source or the white point of image displaying devices such as TV and PC monitor. In this paper, our goal is to find an appropriate method of convert...
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