The standard wavelet transform can be considered to be a subtree of a complete multiresolution binary tree, in which each branch corresponds to a decomposition step. Any subtree of this multiresolution tree which cove...
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The standard wavelet transform can be considered to be a subtree of a complete multiresolution binary tree, in which each branch corresponds to a decomposition step. Any subtree of this multiresolution tree which covers all leaves is a valid wavelet packet processing tree, which specifies the processing steps for a wavelet decomposition of the original data. The present paper describes an adaptation scheme for configuring a wavelet packet processing tree using a genetic algorithm search of the complete multiresolution tree. Data compression is used as the target signalprocessing task. Trees are configured for different compression rates, and performance is measured based on the reconstruction distortion. Performances are validated using testing as well as training data; the performance of the configured trees are compared to that of the standard wavelet transform.
A predictive image coder having minimal decoder complexity is presented. The image coder utilizes recursive interpolative DPCM in conjunction with adaptive classification, entropy-constrained trellis coded quantizatio...
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A predictive image coder having minimal decoder complexity is presented. The image coder utilizes recursive interpolative DPCM in conjunction with adaptive classification, entropy-constrained trellis coded quantization, and optimal rate allocation to obtain signal-to-noise ratios in the range of the most advanced transform coders.< >
Least-squares sine-fit algorithms are used extensively in signalprocessing applications. The parameter estimates produced by such algorithms are subject to both random and systematic errors when the record of input s...
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Least-squares sine-fit algorithms are used extensively in signalprocessing applications. The parameter estimates produced by such algorithms are subject to both random and systematic errors when the record of input samples consists of a fundamental sine wave corrupted by harmonic distortion or noise. The errors occur because, in general, such sine-fits will incorporate a portion of the harmonic distortion or noise into their estimate of the fundamental. Bounds are developed for these errors for least-squares four-parameter (amplitude, frequency, phase, and offset) sine-fit algorithms. The errors are functions of the number of periods in the record, the number of samples in the record, the harmonic order, and fundamental and harmonic amplitudes and phases. The bounds do not apply to cases in which harmonic components become aliased.< >
Several DSP algorithms need to remove high-frequency or impulsive noise while preserving edges, e.g., in speech and image processing applications: median filtering has been proved more effective to achieve this goal t...
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Several DSP algorithms need to remove high-frequency or impulsive noise while preserving edges, e.g., in speech and image processing applications: median filtering has been proved more effective to achieve this goal than other filtering techniques. Efficient architectural implementation for real-time applications involves a careful VLSI design, which takes into account modularity, regularity, adaptability, throughput, circuit complexity, and fault tolerance. Two new architectural approaches are presented in this paper to deal with different application and implementation constraints: the pipelined polarizing median filter and the pipelined sorting median filter.< >
Ultrasonic pulse-echo ranging systems based on threshold detection methods are popular devices in the robotic field, as a means for determining the proximity of objects in a cost-effective manner. Despite their widesp...
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Ultrasonic pulse-echo ranging systems based on threshold detection methods are popular devices in the robotic field, as a means for determining the proximity of objects in a cost-effective manner. Despite their widespread use, serious concerns are often raised regarding the accuracy of the sensed data, particularly when the return signals are received at low signal-to-noise ratios. In principle, correlation-based detection methods provide better performance for their outstanding capability of detecting and recovering weak signals buried in noise, so as to permit ranging at longer distances, or, the distance being the same, at higher frequencies, with resulting improvements in spatial resolution in spite of the increased attenuation. In this paper the authors describe a simple pulse-echo ranging system whose receiver signalprocessing is completely digital: the use of appropriate sampling techniques and signalprocessingalgorithms allows one to benefit from the advantages of correlation-based detection methods for accurate ranging of multiple objects, without compromising the most prominent features of pulse-echo sonar systems, namely their relatively low cost and simplicity of operation. The experimental results presented concern the use of relatively high-frequency ultrasonic transducers.< >
A programmable longitudinal feedback system based on four AT&T 1610 digital signal processors has been developed as a component of the PEP‐II R&D program. This longitudinal quick prototype is a proof of conce...
A programmable longitudinal feedback system based on four AT&T 1610 digital signal processors has been developed as a component of the PEP‐II R&D program. This longitudinal quick prototype is a proof of concept for the PEP‐II system and implements full‐speed bunch‐by‐bunch signalprocessing for storage rings with bunch spacings of 4 ns. The design incorporates a phase‐detector‐based front end that digitizes the oscillation phases of bunches at the 250 MHz crossing rate, four programmable signal processors that compute correction signals, and a 250‐MHz hold buffer/kicker driver stage that applies correction signals back on the beam. The design implements a general‐purpose, table‐driven downsampler that allows the system to be operated at several accelerator facilities. The hardware architecture of the signalprocessing is described, and the software algorithms used in the feedback signal computation are discussed. The system configuration used for tests at the LBL advanced Light Source is presented.
The Wigner-Ville distribution (WD) is particularly used for time-frequency analysis of non-stationary signals. This paper proposes the application of the smoothed version of the WD to design an innovative virtual inst...
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The Wigner-Ville distribution (WD) is particularly used for time-frequency analysis of non-stationary signals. This paper proposes the application of the smoothed version of the WD to design an innovative virtual instrument (VI) which can be used as a spectral analyzer. Our work focuses on the computational algorithms that meet several essential requirements for implementation, such as high time-frequency resolution, precision, flexibility, and computational efficiency. An original interpolation formula is derived, to overcome the negative effects due to leakage. Following this, a simple WD implementation on the designed VI is proposed and a series of numerical checks is shown, to suitably characterize the new instrument.< >
This paper proposes a novel method for detecting the optimal sequence of prosodic phrases from continuous speech based on data-driven approach. The pitch pattern of input speech is divided into prosodic segments which...
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This paper proposes a novel method for detecting the optimal sequence of prosodic phrases from continuous speech based on data-driven approach. The pitch pattern of input speech is divided into prosodic segments which minimized the overall distortion with pitch pattern templates of accent phrases by using the One Pass search algorithm. The pitch pattern templates are designed by clustering a large number of training samples of accent phrases. On the ATR continuous speech database uttered by 10 speakers, the rate of correct segmentation was 91.7% maximum for the same sex data of training and testing, 88.6% for the opposite sex.< >
This paper reports on results from the algorithmic design and simulation of a two-path poly-phase decimation filter with 24-bit accuracy over the frequency range from dc to approximately 16 kHz. The filter is suited f...
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This paper reports on results from the algorithmic design and simulation of a two-path poly-phase decimation filter with 24-bit accuracy over the frequency range from dc to approximately 16 kHz. The filter is suited for very high precision data conversion applications, and has been designed for use with a fourth-order /spl Sigmaspl Delta/ modulator running at 4096 kHz. This paper also reports on the fixed-point architectural design, comparative bit-level simulations and silicon implementation.< >
In this paper, a unified approach towards the integration of sensor data is presented. Using a modified recursive least squares algorithm, variations in parameter location and measurement noise may be tracked. The int...
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In this paper, a unified approach towards the integration of sensor data is presented. Using a modified recursive least squares algorithm, variations in parameter location and measurement noise may be tracked. The introduction of fuzzy logic, and the subsequent development of operators especially suited for sensor integration, enables sensors of radically different nature to be combined in a consistent manner.< >
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