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Acoustics, Speech, and Signal Processing, 2000. ICASSP '00. Proceedings. 2000 IEEE International Conference on

Date 5-9 June 2000

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  • 2000 IEEE International Conference on Acoustics, Speech, and Signal Processing Proceedings [front matter]

    Publication Year: 2000 , Page(s): i - lxxx
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    Freely Available from IEEE
  • Author index

    Publication Year: 2000 , Page(s): A1 - A11
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    Freely Available from IEEE
  • Serially concatenated space-time code for high data rate wireless communication systems

    Publication Year: 2000 , Page(s): 3678 - 3681 vol.6
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    This paper suggests and analyzes the serially concatenated space-time code (SCSTC) with the possibility of a efficient high-speed transmission in a band limited channel. The suggested code has a structure that uses the interleaver to connect the space-time code as a inner code and the convolutional code as an outer code serially. This code keeps the advantage of high-speed transmission and also has a high BER performance. The performance of the suggested system is compared with the conventional bandwidth efficient trellis coded modulation, such as a serially concatenated trellis coded modulation (SCTCM) and a turbo-trellis coded modulation (turbo-TCM). The results show that the suggested system has a 2.8 dB and a 3 dB better BER performance than SCTCM and turbo-TCM respectively in the case of a 2b/s/Hz transmission rate in a fading channel View full abstract»

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  • Efficient CELP-based diversity schemes for VoIP

    Publication Year: 2000 , Page(s): 3682 - 3685 vol.6
    Cited by:  Papers (6)  |  Patents (5)
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    Diversity schemes include information about packet n in future packets or send information about packet n via separate paths. If packet n is lost, it is reconstructed from information included in future packets or information received via separate paths. This paper presents CELP-based diversity schemes for voice over packet applications. The diversity schemes reduce the impact of packet losses while being efficient in terms of both bandwidth requirement and computational complexity. With our diversity schemes, transmission schemes that allocate bandwidth resources among diversity stages during congestion give significantly better performance than schemes that use no diversity during congestion, for the same bandwidth usage View full abstract»

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  • Code-length-based universal extraction for blind signal separation

    Publication Year: 2000 , Page(s): 3422 - 3425 vol.6
    Cited by:  Papers (1)
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    We propose a blind signal separation algorithm (CLUE) that uses the sum of the individual code lengths of the extracted signals as a measure of the separation performance. The new technique combines a widely-available universal data compression routine with any single-parameter search procedure. Unlike previous approaches, the proposed method is model-free and does not rely on the moment values of the signals for its separation performance. An example shows the algorithm's efficiency in separating mixtures of image, audio, and text data View full abstract»

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  • A new approach to model communication for mapping and scheduling DSP-applications

    Publication Year: 2000 , Page(s): 3354 - 3357 vol.6
    Cited by:  Papers (1)
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    We present a novel approach to model inter-processor communication in multi-DSP systems. In most multi-DSP systems, inter-processor communication is realized by transferring data over point-to-point links with hardware FIFO buffers. Direct memory access (DMA) is additionally used to concurrently transfer data to the FIFO buffers and perform computation. Our model accounts for the limited size of the communication buffers as well as concurrent DMA transfer. This novel communication model is applied in our rapid prototyping environment for optimizing multi-DSP systems. Given an extended data flow graph of the DSP application and a description of the target multi-processor system our rapid prototyping environment automatically maps the DSP application onto the multi-processor system and generates a schedule for each processor View full abstract»

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  • Development of a speech processor for laboratory experiments with cochlear implant patients

    Publication Year: 2000 , Page(s): 3626 - 3629 vol.6
    Cited by:  Papers (3)
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    A laboratory speech processor based on Motorola's DSP56002 chip was developed for cochlear implant research. The hardware of the speech processor is described in detail and the safety issues associated with electrical stimulation of the auditory nerve are discussed. The speech processor is capable of providing high-rate stimulation to six electrodes using short biphasic pulses presented either simultaneously or in an interleaved fashion. Different speech processing algorithms including the continuous interleaved sampling (CIS) strategy were implemented in this processor and tested successfully with cochlear implant patients View full abstract»

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  • Issues in high-speed WLANs

    Publication Year: 2000 , Page(s): 3698 - 3701 vol.6
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    This paper presents an overview of the physical layer (PHY) and medium access control layer (MAC) requirements for a new class of high-speed wireless local area networks (WLANs) operating in the 5 GHz Unlicensed National Information Infrastructure (UNII) bands at data rates of 6-54 Mbps View full abstract»

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  • Software-gated pulse-Doppler ultrasound for a DSP-based blood flowmeter

    Publication Year: 2000 , Page(s): 3598 - 3601 vol.6
    Cited by:  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB)  

    Traditional short-gate pulse-Doppler devices rely on hardware gating to measure a blood vessel's velocity profile. In the same spirit as software radio, advanced digital signal processor (DSP) technologies suggest software gating as an alternative to traditional hardware gating methods. By way of computer simulation, this paper explores the viability of a software-gated pulse-Doppler technique for measuring a blood vessel's velocity profile. Simulations utilize a particle model that is then mixed, filtered, and sampled. Spectral analysis provides velocity profile information. Preliminary results suggest software gating is not only feasible but, also advantageous View full abstract»

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  • Wavelet packet transforms for system-on-chip applications

    Publication Year: 2000 , Page(s): 3287 - 3290 vol.6
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    A methodology for the production of silicon cores for wavelet packet decomposition has been developed. The scheme utilizes efficient scalable architectures for both orthonormal and biorthgonal wavelet transforms. The cores produced from these architectures can be readily scaled for any wavelet function and are easily configurable for any subband structure. The cores are fully parameterized in terms of wavelet choice and appropriate wordlengths. Designs produced are portable across a range of silicon foundries as well as FPGA and PLD technologies. A number of exemplar implementations have been produced View full abstract»

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  • Rapid prototyping for mixed architectures

    Publication Year: 2000 , Page(s): 3263 - 3266 vol.6
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    The aim of our work is to achieve a rapid prototyping dedicated to mixed architectures, made up of one multi-DSP part (software) and a FPGA architecture (dedicated hardware). We have determined a complete codesign methodology enabling to implement a complete digital signal or image processing line. The starting description associated with this methodology, is only a functional description, represented by a data flow graph. This global process developed for an automatic implementation enables the user to develop complex applications at a high level onto a complex architecture without any implementation pre-requirements View full abstract»

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  • A novel multiply multiple accumulator component for low power PDSP design

    Publication Year: 2000 , Page(s): 3247 - 3250 vol.6
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (328 KB)  

    This paper presents a novel programmable digital signal processor (PDSP) component called the multiply multiple accumulator (MMAC). The MMAC differs from a standard multiply accumulator (MAC) in that it has k addressable accumulators rather than 1 in the case of the MAC. It is demonstrated that this feature of the MMAC can provide for low power scheduling of FIR filter operations. Typically, the number of read accesses to associated memories can come down, asymptotically, by a factor of k. The switching activity of associated multipliers also comes down by a factor of k View full abstract»

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  • Information-theoretic analysis of watermarking

    Publication Year: 2000 , Page(s): 3630 - 3633 vol.6
    Cited by:  Papers (9)  |  Patents (1)
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    An information-theoretic analysis of watermarking is presented in this paper. We formulate watermarking as a communication problem with side information at the encoder and decoder and determine the hiding capacity, which upper-bounds the rates of reliable transmission and quantifies the fundamental tradeoff between three quantities: the achievable watermarking rates and the allowed distortion levels for the information hider and the attacker. The hiding capacity is the value of a game between the information hider and the attacker. The optimal attack strategy is the solution of a particular rate-distortion problem, and the optimal hiding strategy is the solution to a channel coding problem. For several important problems, the hiding capacity is the same whether or not the decoder knows the host data set. It is also shown that existing watermarking systems in the literature operate far below capacity View full abstract»

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  • Implementation options for WCDMA

    Publication Year: 2000 , Page(s): 3702 - 3705 vol.6
    Cited by:  Papers (1)  |  Patents (5)
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    This paper discusses the design tradeoffs and implementation options for building wideband CDMA systems. System on a chip (SOC) solutions have a range of implementation options, from processor cores to custom ASIC (or a mix of both), to satisfy the extremely challenging requirements of the digital baseband of a next generation wireless system. Additionally, good designs should have a roadmap that will take advantage of rapid improvements in process technology. This paper describes the use of coprocessors alongside DSPs to meet the demanding computational requirements of WCDMA. We show that a coprocessor based design still achieves reasonably efficient area and power use, maintains a high degree of programmability and has a good technology migration path View full abstract»

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  • NMR signal enhancement via a new time-frequency transform

    Publication Year: 2000 , Page(s): 3602 - 3605 vol.6
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    A reliable method to reduce the noise from nuclear magnetic resonance (NMR) signals using a previously developed linear critically-sampled time-frequency transform is proposed. In addition to its low computational requirements, this transform has many theoretical advantages that make it a good candidate for NMR signal enhancement. NMR signals, in the transform domain, are concentrated in a very few number of coefficients while the noise is fairly distributed among the coefficients. Therefore, performing a thresholding technique in the transform domain significantly enhances the signal. Comparison with other noise reduction techniques used for the same purpose showed that this technique has superior performance thus confirming the theoretical expectations View full abstract»

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  • Robust clustering of acoustic emission signals using the Kohonen network

    Publication Year: 2000 , Page(s): 3891 - 3894 vol.6
    Cited by:  Papers (1)
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    Acoustic emission-based techniques are promising for nondestructive inspection of mechanical systems. For reliable automatic fault monitoring, it is important to identify the transient crack-related signals in the presence of strong time-varying noise and other interference. In this paper we propose the application of the Kohonen network for this purpose. The principal components of the short-time Fourier transforms of the data were applied input of the network. The clustering results confirm the capability of the Kohonen network for reliable source identification of acoustic emission signals, assuming enough care has been taken in implementing the training algorithm of the network View full abstract»

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  • Discriminative training for large vocabulary telephone-based name recognition

    Publication Year: 2000 , Page(s): 3739 - 3742 vol.6
    Cited by:  Papers (4)
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    This paper describes progress on a commercial application of the MECS recognition system to the task of recognizing Japanese family names spoken by customers into the answering machines of a large marketing/human resource company. The task is thus speaker-independent, open vocabulary, and is characterized by large variation in caller speaking styles, telephone types and acoustic environments. Our results show that context-independent hidden Markov models trained discriminatively with the minimum classification error criterion are a practical alternative to context-dependent models based on phonetic decision trees, yielding better performance with a much smaller number of parameters. On this difficult task we have obtained 59% correct family name recognition. A phoneme-based confidence measure enables us to obtain 85% correct name recognition for accepted utterances, at an overall utterance acceptance rate of 15% View full abstract»

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  • Linear prediction for bandpass signals based on nonuniform past samples

    Publication Year: 2000 , Page(s): 3854 - 3857 vol.6
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (312 KB)  

    This paper concerns linear prediction of the value of a bandpass signal containing one or more passbands from a finite set of its past samples. The method of choosing prediction coefficients involves the eigenvector corresponding to the smallest eigenvalue of a matrix dependent on a function which is the Fourier transform of the set of intervals making up the passband. The method is developed for a set of arbitrary past samples and applied here to a set of “interlaced” samples that are nonuniform but periodic. The method applies to finite energy signals as well as to bandpass signals of polynomial growth, which connects to the theory of generalized functions. Computational examples are given of prediction coefficient values and of signal predictions View full abstract»

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  • Rejection based on a posteriori probability estimated by MLP with application for Mandarin voice dialer on ASIC

    Publication Year: 2000 , Page(s): 3446 - 3449 vol.6
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    High performance Mandarin voice dialer is much more difficult than its English counterpart to achieve, especially on inexpensive hardware as ASIC. One way to improve its performance is to incorporate rejecters into the system. In our study, an MLP based postprocessor, an a posteriori probability estimator, is applied after HMM Viterbi recognition. Poor utterances, which are recognized by HMMs but have low a posteriori probability, are then rejected. Rejecting 4.9% of all the testing utterances, the MLP rejector boosts the HMM-based system's single digit accuracy from 97.1% to 99.6% for the Mandarin voice dialer, a ten-syllable speaker independent task. The performance is better than those of rejection based on linear discrimination, anti-digit models or likelihood ratio View full abstract»

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  • Time-frequency methods for analyzing structural response data

    Publication Year: 2000 , Page(s): 3878 - 3881 vol.6
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    This paper considers the application of various time-frequency methods to structural response data. It illustrates that by use of time-frequency methods information that is not evident in impulse or frequency responses data is brought to light. The shortcomings of classical bilinear time-frequency representations are discussed. A selection of data adaptive time-frequency methods designed as enhancements to the classical techniques are compared and contrasted for this application. It is shown that many of these algorithms fail to give acceptable results when applied to the comparatively complex data sets considered here View full abstract»

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  • Source adaptive software 2D iDCT with SIMD

    Publication Year: 2000 , Page(s): 3642 - 3645 vol.6
    Cited by:  Papers (1)  |  Patents (2)
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    This paper presents a fast two-dimensional inverse discrete cosine transform that adapts to compressed video source statistics to reduce execution time. iDCT algorithms for sparse blocks eliminate calculations for some zero coefficients and are implemented with quad-word parallel single-instruction-multiple-data (SIMD) multimedia instructions. It is observed that end-of-block marker value histograms vary little within single shots. An adaptive control mechanism is proposed that chooses the optimal set of iDCTs to prepare for an entire shot from its 1st frames (to reduce software overheads and penalties). This introduces no degradation of decoded video quality compared with a conventional SIMD 8×8 iDCT implemented with Intel MMX instructions. It is confirmed that execution time is reduced an additional 15% with Murata's method for 4 Mbps MPEG2 natural video. In comparison, execution time is reduced 22% with a modified version Murata's method, and by 35% with the new source adaptive method View full abstract»

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  • Implementation of a hands-free car phone with echo cancellation and noise-dependent loss control

    Publication Year: 2000 , Page(s): 3622 - 3625 vol.6
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    An implementation of a complete hands-free telephone system on a low-cost DSP with 16-bit fixed-point arithmetic is presented. The prototype, consisting of an adaptive echo canceller, was developed for an industrial customer. The main emphasis is laid on the control unit of the system being necessary for a reliable performance in all environments and the attenuation introduced by the loss control unit. Keeping the loss as small as possible, which is most important for a natural communication including full double-talk capabilities, requires exact knowledge of the current system performance and the background noise level. In case of background noise the applied total loss is adjusted to the power of the noise. The measures we used and their application to our system control are discussed View full abstract»

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  • Neural network boundary refining for automatic speech segmentation

    Publication Year: 2000 , Page(s): 3438 - 3441 vol.6
    Cited by:  Papers (9)  |  Patents (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    This work is an extension of a previous work in which an automatic speech segmentation and labeling system was proposed based on a hidden Markov model (HMM) speech recognizer followed by a fuzzy-logic boundary correction system. In this paper we explore the possibility of substituting that difficult to design fuzzy-logic system by a neural network (NN) based system that can be automatically trained. First, the whole fuzzy-logic boundary correction system, which used different rule sets for each kind of phonetic transition, has been substituted by a single NN. Results show that this single NN outperforms the complete fuzzy-logic system. Then, the possibility of using different NNs specialized in each kind of phonetic transition has been explored. Results are again clearly better than the results obtained with the fuzzy-logic system, but not clearly better than the results obtained with just one NN View full abstract»

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  • Smart antenna receiver for GSM/DCS system based on single chip solution

    Publication Year: 2000 , Page(s): 3207 - 3210 vol.6
    Cited by:  Papers (1)
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    This paper presents a single chip implementation of a space-time algorithm for co-channel interference (CCI) and intersymbol interference (ISI) reduction in GSM/DCS systems. The temporal channel for the Viterbi receiver and the beamformer weights for the CCI rejection are estimated jointly by optimizing a suitable cost function for separable space-time channels. By taking into account of the current integration capabilities provided by the FPGA (field programmable gate array), the feasibility of a single chip JSTE solution based on a three-processor architecture for carrier beamforming, equalization and demodulation is demonstrated View full abstract»

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  • Modeling text with generalizable Gaussian mixtures

    Publication Year: 2000 , Page(s): 3494 - 3497 vol.6
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (324 KB)  

    We apply and discuss generalizable Gaussian mixture (GGM) models for text mining. The model automatically adapts model complexity for a given text representation. We show that the generalizability of these models depends on the dimensionality of the representation and the sample size. We discuss the relation between supervised and unsupervised learning in the test data. Finally, we implement a novelty detector based on the density model View full abstract»

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