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Conference on advanced signal processing algorithms, architectures, and implementations XI

作者： Pearlman, WA Medley, MJ Suter, BW Rensselaer Polytech Inst Ctr Next Generat Video Res Troy NY 12180 USA

ISBN: (纸本)0819441880

Compressed video bitstreams require protection from channel errors in a wireless channel. The three-dimensional (3-D) SPIHT coder has proved its efficiency and its real-time capability in compression of video. A forward-error-correcting (FEC) channel (RCPC) code combined with a single ARQ (automatic-repeat-request) proved to be an effective means for protecting the bitstream. In this paper, the need for ARQ is eliminated by making the 3-D SPIHT bitstream more robust and resistant to channel errors. Packetization of the bitstream and the reorganization of these packets to achieve scalability in bit rate and/or resolution in addition to robustness is demonstrated and combined with channel coding to not only protect the integrity of the packets, but also allow detection of packet decoding failures, so that only the cleanly recovered packets are reconstructed. In extensive comparative tests, the reconstructed video is shown to be superior to that of MPEG-2, with the margin of superiority growing substantially as the channel becomes noisier.

关键词： video compression video transmission robust source coding error resilient transmission combined source-channel coding 3-D wavelet transform embedded wavelet coding

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Low Power implementation of a Turbo-decoder on programmable architectures

Low Power implementation of a Turbo-decoder on programmable ...

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Asia and South Pacific Design Automation Conference (ASP-DAC 2001)

作者： Gilbert, F Worm, A Wehn, N Univ Kaiserslautern Dept Elect Engn & Informat Technol Inst Microelect Syst D-67663 Kaiserslautern Germany

ISBN: (纸本)0780366336

Low Power is an extremely important issue for future mobile radio systems. Channel decoders are essential building blocks of base-band signal processing units in mobile terminal architectures. Thus low power implementations of advanced channel decoding techniques are mandatory. In this paper we present a low power implementation of the most sophisticated channel decoding algorithm (Turbo-decoding) on programmable architectures. Low power optimization is performed on two abstraction levels: on system level by the use of an intelligent cancellation technique, on implementation level by the use of dynamic voltage scaling. With these techniques we can reduce the worst case energy consumption to 55% using data of state-of-the-art processors. Our approach is also applicable for hardware implementations. To the best of our knowledge, this is the first in-depth study of low power implementations of Turbo-decoders based on voltage scheduling for third generation wireless systems.

关键词： Bit error rate Channel coding Dynamic voltage scaling Error correction Forward error correction Iterative decoding Processor scheduling Redundancy signal processing algorithms Turbo codes

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Development of transient power quality indices based on time-frequency distribution

Development of transient power quality indices based on time...

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advanced signal processing algorithms, architectures, and implementations Xv

作者： Shin, Yong-June Crapse, Philip Department of Electrical Engineering University of South Carolina Columbia SC 29208 United States

For an assessment of the power quality in power distribution systems, classical Fourier series-based power quality indices are normally employed. The classical Fourier series-based power quality indices assume the periodicity of the disturbance so that the applications are limited to the harmonics. Hence, it is necessary for us to redefine power quality indices for the "transient" disturbances. In this paper, development of time-frequency based power quality indices are discussed for an assessment of transient power quality. The time and frequency localized information of the transient disturbance signals will be utilized for a new definition of the transient power quality indices. As an example of time-frequency based power quality indices, new definition of transient telephone interference factor has been carefully derived and verified in comparison with traditional telephone interference factor has been carefully derived and verified in comparison with traditional telephone interference factor. Time-frequency based power quality indices allow one to quantify the effects of transient disturbances by time and frequency localized information.

关键词： signal processing

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Bandlimits and coherent modulation filtering of non-stationary signals

Bandlimits and coherent modulation filtering of non-stationa...

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Conference on advanced signal processing algorithms, architectures, and Implementantions XvIII

作者： Clark, Pascal Atlas, Les Univ Washington Dept Elect Engn Seattle WA 98195 USA

ISBN: (纸本)9780819472946

Modulation filtering is a technique for filtering slowly-varying envelopes of frequency subbands of a non-stationary signal, ideally without affecting the signal's phase and fine-structure. Coherent modulation filtering is a potentially more effective subtype of such techniques where subband envelopes are determined through demodulation of the subband signal with a coherently detected subband carrier. In this paper we propose a coherent modulation filtering technique based on detecting the instantaneous frequency of a subband from its time-frequency representation. We show that coherent modulation filtering imposes a new bandlimiting constraint on the modulation product plus the ability to recover arbitrarily chosen envelopes and carriers from their modulation product. We show that a carrier estimate based on the time-varying spectral center-of-gravity satisfies the bandlimiting condition as well as Loughlin's previously derived bandlimiting constraint on the instantaneous frequency of carrier. These bandwidth constraints lead to effective and distortion-free modulation filters, offering new approaches for potential signal modification. The spectral center-of-gravity does not satisfy the condition on arbitrary recovery, however, which somewhat limits the flexibility of coherent modulation filtering. Demonstrations are provided with speech signals.

关键词： modulation time-frequency analysis time-varying filters bandwidth speech processing

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BIT-SERIAL architectures FOR PARAMETER EXTRACTION AND PHONETIC ENCODING

BIT-SERIAL ARCHITECTURES FOR PARAMETER EXTRACTION AND PHONET...

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5TH ANNUAL EUROPEAN COMPUTER CONF : advanced COMPUTER TECHNOLOGY, RELIABLE SYSTEMS AND APPLICATIONS

作者： MARTINEZ, E MARTINEZ, P RODELLAR, v GOMEZ, P

ISBN: (纸本)0818621419

One interesting application of digital signal processing is low-level speech recognition. A description is presented of the design of a whole structure made up of two substructures: an LPC (linear predictive coding) filter to compact the information present in the voice trace and a bit serial neural network which is used for the phonetic encoding of this trace. These substructures are designed with a silicon compiler adapted for digital signal algorithms and bit-serial architectures.

关键词： signal processing

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Modeling multidimensional data and control flow

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IEEE Transactions on very Large Scale Integration (vLSI) Systems 1993年第3期1卷 319-326页

作者： Franssen, Frank H.M. Balasa, Florin van Swaaij, Michael F.X.B. Catthoor, Francky v.M. De Man, Hugo J. IMEC Leuven Belgium

Multidimensional (M-D) signal processing is a key component of most real-time signal and data processing vLSI systems in industry. Handling the M-D nature of data in an efficient way is crucial to arrive at acceptable system implementations. This aspect of design has often been ignored up to now in high-level synthesis. In this paper, a polyhedral based model is presented for the linear, piecewise linear and data dependent signal indexing as occurring in practical M-D signal processing applications. The model features a mathematical description of dependencies between individual operations and signal instances of M-D signals for all algorithms that contain signal indexing specified by a mixture of indexing by iterators and M-D signals embedded in the data flow. The exact modeling of M-D signal indexing is especially essential for deriving alternative control flow structures for a given data flow specification. Exploration of various control flow structures allows to arrive at an efficient large-scale memory organization during high level synthesis of architectures, both in terms of storage locations and access order.

关键词： signal processing

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Subband-domain signal processing for radar array systems

Subband-domain signal processing for radar array systems

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Proceedings of the 1999 advanced signal processing algorithms, architectures, and implementations IX

作者： Rabinkin, Daniel v. Pulsone, Nicholas B. MIT Lincoln Lab Lexington United States

Subband-domain algorithms provide an attractive technique for wideband radar array processing. The subband-domain approach decomposes a received wideband signal into a set of narrowband signals. While the number of processing threads in the system increases, the narrowband signals within each subband can be sampled at a correspondingly slower rate. Therefore, the data rate at the input is similar to that at the output of the subband processor. There are several advantages to the subbanding method. It can simplify typical radar algorithms such as adaptive beamforming and equalization by the virtue of reducing subband signal bandwidth, thereby potentially reducing the computational complexity over an equivalent tapped-delay line approach. It also allows for a greater parallelization of the processing task, hence enabling the use of slower and less power consuming hardware. In order to evaluate the validity of the subbanding approach, it is compared with conventional processing methods. This paper focuses on adaptive beamforming and pulse compression performance for a wideband radar system. The performance of an adaptive beamformer is given for a polyphase filter based subband approach and is measured against narrowband processing. SINR loss curves and beampatterns for a subband system are presented. Design criteria for subband polyphase filter processing that minimizes signal distortion are provided and the distortion is characterized. Finally subband-domain pulse compression is demonstrated and compared with the conventional approach.

关键词： Radar systems

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New one-hot RNS structures for high-speed signal processing

New one-hot RNS structures for high-speed signal processing

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Conference on advanced signal processing algorithms, architectures and implementations XII

作者： Conway, R Conway, T Nelson, J Univ Limerick ECE Dept Limerick Ireland

ISBN: (纸本)0819445584

New efficient structures using the one-hot residue number system (OHRNS) are presented. Normally the RNS uses a binary representation for the residues, though recently there has been renewed interest in the OHRNS, which uses a simple, but novel representation for the residues. The basic component of the OHRNS is the barrel shifter, making the OHRNS suitable for very high speed applications. The first of the new structures presented reduces the power dissipation in OHRNS adder trees. A modification to the normal barrel shifter is proposed, which reduces the power dissipated by as much as 30%. This improvement is obtained through the use of the modified barrel shifter and the appropriate connection of active-low and active-high stages. This overall power reduction offers the possibility of using the OHRNS in place of a typical full adder based tree in high speed DSP applications. A new storage register for one-hot representations is detailed, which overcomes the problem of having to use a large number of registers. A new architecture is presented for fast OHRNS sign detection. Sign detection is complex and slow to perform in the RNS. A mixed radix conversion (MRC) is typically used for sign detection in the OHRNS. The new sign detection architecture is based on a new property of the Chinese Remainder Theorem (CRT) and is significantly faster than the MRC approach for large moduli sets. Simulation results using SPICE are detailed for the new structures.

关键词： RNS one-hot sign detection

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Damped harmonics and polynomial phase signals

Damped harmonics and polynomial phase signals

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advanced signal processing: algorithms, architectures, and implementations v

作者： Zhou, Guotong Giannakis, Georgios B. Univ. of Virginia Charlottesville VA USA

ISBN: (纸本)0819416207

The concern here is of retrieving damped harmonics and polynomial phase signals in the presence of additive noise. The damping function is not limited to the exponential model, and in certain cases, the additive noise does not have to be white. Three classes of algorithms are presented, namely DFT based, Kumaresan-Tufts type extensions, and subspace variants including the MUSIC algorithm. Preference should be based on the available data length and frequency separations. In addition, retrieval of self coupled damped harmonics, which may be present when nonlinearities exist in physical systems, is investigated. Simulation examples illustrate main points of the paper.

关键词： signal processing

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Frequency-selective techniques based on singular value decomposition (SvD), total least squares (TLS), and bandpass filtering

Frequency-selective techniques based on singular value decom...

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advanced signal processing: algorithms, architectures, and implementations v

作者： Chen, Hua van Huffel, Sabine vandewalle, Joos Katholieke Univ. Leuven Leuven Belg

ISBN: (纸本)0819416207

Much research has focused on estimating the parameters of a sum of exponentially damped sinusoids. In some applications, such as nuclear magnetic resonance signal processing, only a few of the sinusoids are of interest. This paper presents some new frequency-selective techniques for estimating the parameters of sinusoids within a specified frequency region using a subspace and SvD-based estimation algorithm and an FIR filter matrix. The applicable estimation algorithms in this technique are the linear prediction method, the matrix pencil method, Kung et al.'s method and its total-least-squares variant, called the HTLS method. The benefits of the frequency-selective technique combined with the HTLS estimation method are confirmed through simulations.

关键词： signal processing

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