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A space-time coded transmitter diversity technique for frequency selective fading channels

A space-time coded transmitter diversity technique for frequ...

引用

Sensor Array and Multichannel signal processing Workshop

作者： K.F. Lee D.B. Williams Motorola Inc. Boynton Beach FL USA Center for Signal and Image Processing School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

A simple space-time coded orthogonal frequency division multiplexing (OFDM) transmitter diversity technique for wireless communications over frequency selective fading channels is presented. The proposed technique utilizes OFDM to transform frequency selective fading channels into multiple flat fading subchannels on which space-time coding is applied. A two-branch transmitter diversity system is implemented without bandwidth expansion and with a small increase in complexity beyond that of a conventional OFDM system. Simulations verify that the two-branch transmitter diversity system achieves a diversity gain equivalent to that of the optimal maximal ratio combining (MRC) receiver diversity system.

关键词： Transmitters Frequency diversity Fading Diversity reception Delay effects OFDM modulation Discrete Fourier transforms Wireless communication Base stations Receiving antennas

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Unsupervised estimation of the human vocal tract length over sentence level utterances

Unsupervised estimation of the human vocal tract length over...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： B.F. Necioglu M.A. Clements T.P. Barnwell Sch. of Electr. & Comput. Eng. Georgia Inst. of Technol. Atlanta GA USA Center for Signal and Image Processing School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta CA USA

This paper describes a method for the unsupervised and gender-independent estimation of the average human vocal tract length from the speech waveform, and reports results obtained on Fant's (1960) X-ray vowel data as well as results from experiments performed on multiple sentence utterances of 86 male and 78 female TIMIT speakers, including correlation analyses between the vocal tract length estimates and given body heights. The investigated error criteria that make non-iterative, closed-form estimator solutions possible are all found to achieve good speaker clustering potential for both male and female subgroups.

关键词： Human voice Impedance Poles and zeros Frequency estimation Speech analysis Shape measurement signal processing image processing Data engineering Speech processing

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Repackaging signals, systems and circuits in the core ECE curriculum

Repackaging signals, systems and circuits in the core ECE cu...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： T.P. Barnwell J.H. McClellan R.M. Mersereau R.W. Schafer Center for Signal & Image Processing School of Electrical & Computer Engineering Georgia Institute of Technology Atlanta GA USA

For several years, we have been teaching DSP as a first course in electrical and computer engineering at Georgia Tech. Such a dramatic rearrangement of the introductory material requires a new organization of topics and courses when teaching circuits and systems. In addition, the use of computer-enhanced course materials has a profound impact on the systems courses, which are quite mathematical and abstract in nature. This paper addresses some of the issues encountered when adopting a signal processing first approach.

关键词： Circuits computer science education Educational technology Educational programs Educational institutions signal processing Digital signal processing computer aided instruction Switches image processing

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Data efficient implementation of UWBWA SAR algorithms

Data efficient implementation of UWBWA SAR algorithms

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： R. Rau J.H. McClellan Center for Signal and Image Processing School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

It is shown that the particular form of the frequency support of raw data and focused imagery obtained from an ultra-wideband, wide beamwidth synthetic aperture radar system can be exploited in nonseparable sampling schemes to reduce the overall amount of raw data samples and image pixels that need to be stored and computed. Furthermore, it is demonstrated that the constant integration angle backprojection (CIAB) image former implicitly applies a fan filter that interpolates raw data sampled on a quincunx grid back onto the underlying rectangular grid. This subtle property of the CIAB has not been exploited so far. It leads to higher quality images with less computational complexity.

关键词： Radar antennas Frequency Synthetic aperture radar Radar scattering image sampling Bandwidth Grid computing Focusing Ultra wideband technology Geometry

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Audio signal noise reduction using multi-resolution sinusoidal modeling

Audio signal noise reduction using multi-resolution sinusoid...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： D.V. Anderson M.A. Clements Center for Signal and Image Processing School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

The sinusoidal transform (ST) provides a sparse representation for speech signals by utilizing several psychoacoustic phenomena. It is well suited to applications in signal enhancement because the signal is represented in a parametric manner that is easy to manipulate. The multi-resolution sinusoidal transform (MRST) has the additional advantage that it is both particularly well suited to typical speech signals and well matched to the human auditory system. The currently reported work discusses the removal of noise from a noisy signal by applying an adaptive Wiener filter to the MRST parameters and then conditioning the parameters to eliminate "musical noise". In informal tests MRST based noise reduction was found to reduce background noise significantly better than traditional Wiener filtering and to virtually eliminate the "musical noise" often associated with Wiener filtering.

关键词： Noise reduction Wiener filter Background noise signal processing Acoustic noise Discrete transforms signal analysis Frequency estimation Additive noise signal resolution

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An embedded HMM-based approach for face detection and recognition

An embedded HMM-based approach for face detection and recogn...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： A.V. Nefian M.H. Hayes School of Electrical and Computer Engineering Center for Signal and Image Processing Georgia Institute of Technology Atlanta GA USA

We describe an embedded hidden Markov model (HMM)-based approach for face detection and recognition that uses an efficient set of observation vectors obtained from the 2D-DCT coefficients. The embedded HMM can model the two dimensional data better than the one-dimensional HMM and is computationally less complex than the two-dimensional HMM. This model is appropriate for face images since it exploits an important facial characteristic: frontal faces preserve the same structure of "super states" from top to bottom, and also the same left-to-right structure of "states" inside each of these "super states".

关键词： Hidden Markov models Face detection Face recognition Embedded computing Density functional theory image recognition signal processing image processing Layout image databases

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An algorithm for compression of wideband diverse speech and audio signals

An algorithm for compression of wideband diverse speech and ...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： T.R. Trinkaus M.A. Clements Center for Signal and Image Processing School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

A compression scheme for diverse speech and audio signals is proposed. In this scheme, signals are analyzed with a 2-band QMF filter bank followed by the application of a modulated lapped biorthogonal transform (MLBT) to each of the filter bank channels. Subsequent encoding of transform coefficients is performed using Laplacian optimized scalar and vector quantizers, whose rates are determined by an estimated noise threshold, i.e., masking threshold. Listening tests show that the coder achieves a quality at 32 kbits/s that is preferred over the ITU G.722 coder at 64 kbits/s, for speech, music, and more diverse signals consisting of speech in the presence of eventful background sounds. Both the delay of the coder, at 40 ms, and the level of complexity are moderate.

关键词： Wideband Speech Filter bank signal analysis Laplace equations Acoustic noise Masking threshold Acoustic testing Multiple signal classification Music

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A new directional filter bank for image analysis and classification

A new directional filter bank for image analysis and classif...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： Sang-Il Park M.J.T. Smith R.M. Mersereau Center for Signal and Image Processing School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

A new directional filter bank for image analysis and classification is proposed. This paper introduces an improved structure in order to visualize subband outputs of the directional filter banks, while retaining the attractive properties of the original directional filter banks such as 1-D separable filtering, perfect reconstruction, and maximal decimation. Using this structure, any arbitrary 2/sup n/ band directional filter bank can be implemented by cascading simple directional filter bank blocks, unlike the original structure that needs a parallel structure for visualizing subband outputs. Also, in order to have nondistorted phase information in the subbands for visualization, both FIR and IIR filter prototypes that can be implemented efficiently are provided for linear phase filtering. This paper shows the approach proposed here can be applied to image analysis and classification.

关键词： Filter bank image analysis Visualization Finite impulse response filter IIR filters image reconstruction Prototypes Information filtering Information filters Nonlinear filters

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A wavelet based stereo image coding algorithm

A wavelet based stereo image coding algorithm

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： Qin Jiang Joon Jae Lee M.H. Hayes Center for Signal and Image Processing School of Electrical & Computer Engineering Georgia Institute of Technology Atlanta GA USA Department of Computer Engineering Dong Seo University busan South Korea

Stereo image pair coding is an important issue in stereo data compression. A wavelet based stereo image pair coding algorithm is proposed in this paper. The wavelet transform is used to decompose the image into an approximation image and three edge images. In the wavelet domain, a disparity estimation technique is developed to estimate the disparity field using both approximation image and edge images. To improve the accuracy of estimation of wavelet images produced by the disparity compensation technique, a novel wavelet based subspace projection technique (SPT) is developed. In the SPT, the block dependent subspaces are constructed using block varying basis vectors that are derived from the disparity compensated wavelet images. Experimental results show that the proposed algorithm is efficient to achieve stereo image compression.

关键词： image coding Wavelet domain Wavelet transforms Layout image storage Decoding Wavelet analysis signal processing image processing Data compression

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User interaction in region-based color image segmentation 3rd

User interaction in region-based color image segmentation

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3rd International Conference on Visual Information Systems, VISUAL 1999

作者： Ikonomakis, Nicolaos Plataniotis, Kostas N. Venetsanopoulos, Anastasios N. Department of Electrical and Computer Engineering Digital Signal and Image Processing Lab University of Toronto 10 King’s College Road TorontoONM5S 3G4 Canada School of Computer Science Ryerson Polytechnic University 350 Victoria Street TorontoONM5B 2K3 Canada

ISBN: (纸本)3540660798

An interactive color image segmentation technique is presented for use in applications where the segmented regions correspond to meaningful objects, such as for image retrieval. The proposed technique utilizes the perceptual HSI (hue,saturation,intensity) color space. The scheme incorporates user interaction so that the best possible results can be achieved. Interaction with the user allows the segmentation algorithm to start efficiently and to refine the results. Interaction is performed for the key images (usually the first image or those where new ojects enter the scene) of the video-telephony sequence. The user is allowed to identify on the screen the relevant regions by marking their seeds. The user guidance can be given by the sender, by the receiver, or by both. The effectiveness of the algorithm is found to be much improved over techniques used in the past. © Springer-Verlag Berlin Heidelberg 1999.

关键词： image retrieval

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