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检索条件"主题词=Linear predictive coding"

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Low power showdown: comparison of five DSP platforms implementing an LPC speech codec

Low power showdown: comparison of five DSP platforms impleme...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： D. Hwang C. Mittelsteadt I. Verbauwhede Electrical Engineering Department University of California Los Angeles Los Angeles CA USA

An identical LPC speech coder has been implemented on a set of signal processing specific implementation platforms. The main goal of this experiment was to compare energy consumption. In addition, area/memory requirements and design time are also compared. The coder was first designed in floating-point C. Then, the fixed-point wordlengths were determined. Depending on the platform, either compiled code was generated, assembly code written or a Verilog/VHDL design was created. The platforms reported in this paper include the DSP processors TI C55/spl times/, TI C54/spl times/, TI C6/spl times/ and the design environments Ocapi and A|RT Designer. Energy consumption ranged from 2 /spl mu/J to 288 /spl mu/J per speech frame. Upon scaling the results to the same technology, our results indicated that the lowest power DSP processor (TI C55/spl times/) still consumes a factor of four more energy than an application specific processor.

关键词： Digital signal processing linear predictive coding Speech codecs Signal processing Energy consumption Autocorrelation Speech processing Assembly Hardware design languages Application specific processors

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A study on perceptual distance measure for phase spectrum of stimuli

A study on perceptual distance measure for phase spectrum of...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： H. Banno K. Takeda F. Itakura Graduate School of Engineering University of Nagoya Chikusa Japan Center for Information Media Studies University of Nagoya Chikusa Japan

This paper describes a perceptual distance measure for phase spectrum based on results from a subjective experiment using stimuli. The stimuli have flat amplitude spectrum and, in a particular frequency band, have a certain group delay value. The experiment was performed using stimuli with different group delay peak values where the group delay center frequencies are fixed, and their associated group delay bandwidths are also fixed. It was found that when the peak values of stimuli are between -1 ms and 2 ms, they are perceived to be zero phase regardless of their center frequencies and bandwidths. Moreover, when the peak values are less than -8 ms or more than 10 ms and the bandwidths are less than 1 ERB, each of the stimuli are perceived to be similar. Based on these perceptual similarity results, we introduce an ellipsoidal function to estimate the similarity scores with a simple equation. It is found that the estimated similarity scores well approximate the subjective similarity scores.

关键词： Phase measurement Speech synthesis Delay linear predictive coding Humans Speech coding Bandwidth Frequency Bit rate Speech analysis

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Double-gate MOSFETs: performance and technology options

Double-gate MOSFETs: performance and technology options

引用

International Symposium Semiconductor Device Research (ISDRS)

作者： S. Cristoloveanu F. Allibert A. Zaslavsky LPCS (UMR CNRS and INPG) ENSERG Grenoble France Brown University Providence RI USA

ISBN: (纸本)0780374320

The advantages of double-gate (DG) SOI MOSFETs over conventional, single-gate transistors are described in terms of performance and potential for ultimate scaling. The peculiarity of DG-MOSFETs is that the top and bottom gates are biased simultaneously to establish equal surface potentials: V/sub G2/ = V/sub G1/ for identical gate oxides, or V/sub G2/ = V/sub G1/(t/sub 0X2//t/sub 0X1/) to compensate for the difference in front and back oxide thickness. In fully depleted transistors with a thin enough film, controlling the channel from both sides, forces most of the carriers to flow in the middle of the film, according to the volume inversion concept. Volume inversion results in excellent properties, which will be reviewed in this paper. In particular, the carrier mobility is enhanced, so that the transconductance in double-gate mode exceeds twice the value observed in single-gate mode. A DGMOSFET is more than the sum of two classical transistors.

关键词： MOSFETs Biomembranes Semiconductor films Etching Transconductance Fabrication Oxidation Wafer bonding linear predictive coding Force control

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Comparison of linear prediction cepstrum coefficients and mel-frequency cepstrum coefficients for language identification

Comparison of linear prediction cepstrum coefficients and me...

引用

International Symposium on Intelligent Multimedia, Video and Speech Processing

作者： E. Wong S. Sridharan Speech Research Lab RCSAVT School of Electrical and Electronic Systems Engineering Queensland University of Technology Australia

ISBN: (纸本)9628576623

The speech parametrization methods: linear prediction cepstrum coefficients and mel-frequency cepstrum coefficients were compared with regard to language identification accuracy in a Gaussian mixture model based language identification system. Ten different languages were used to test against a set of ten second test files. The 12th order linear prediction cepstrum coefficients with delta and accelerate coefficients resulted in the best accuracy of 60.0 percent. This has shown that information obtained from linear prediction analysis has increased the ability of discriminating different languages. It also shows that language identification performance may be increased by encompassing temporal information by including delta and acceleration features. Besides, the performance of our test system has proved the feasibility of the modeling language by a single Gaussian Mixture Model instead of using complex system such as phonetic recogniser followed by language modelling or large vocabulary continuous speech recognition system.

关键词： Cepstrum Natural languages linear predictive coding System testing Speech recognition Band pass filters Acceleration Robustness Mel frequency cepstral coefficient Systems engineering and theory

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Distance measures for effective clustering of ARIMA time-series

引用

IEEE International Conference on Data Mining (ICDM)

作者： K. Kalpakis D. Gada V. Puttagunta CSEE Department UMBC Baltimore MD USA

ISBN: (纸本)0769511198

Much environmental and socioeconomic time-series data can be adequately modeled using autoregressive integrated moving average (ARIMA) models. We call such time series "ARIMA time series". We propose the use of the linear predictive coding (LPC) cepstrum for clustering ARIMA time series, by using the Euclidean distance between the LPC cepstra of two time series as their dissimilarity measure. We demonstrate that LPC cepstral coefficients have the desired features for accurate clustering and efficient indexing of ARIMA time series. For example, just a few LPC cepstral coefficients are sufficient in order to discriminate between time series that are modeled by different ARIMA models. In fact, this approach requires fewer coefficients than traditional approaches, such as DFT (discrete Fourier transform) and DWT (discrete wavelet transform). The proposed distance measure can be used for measuring the similarity between different ARIMA models as well. We cluster ARIMA time series using the "partition around medoids" method with various similarity measures. We present experimental results demonstrating that, using the proposed measure, we achieve significantly better clusterings of ARIMA time series data as compared to clusterings obtained by using other traditional similarity measures, such as DFT, DWT, PCA (principal component analysis), etc. Experiments were performed both on simulated and real data.

关键词： linear predictive coding Time measurement Discrete Fourier transforms Discrete wavelet transforms Cepstral analysis Principal component analysis Cepstrum Euclidean distance Indexing Fourier transforms

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An adaptive voice playout method for VOP applications

An adaptive voice playout method for VOP applications

引用

IEEE Conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： A.K. Anandakumar A. McCree E. Paksoy DSP Solutions Research and Development Center Texas Instrumenits Inc. Dallas TX USA

We propose an adaptive voice playout method for handling network delay jitter in voice over packet (VOP) receivers. Our method allows playout delay increase during both silence periods and active speech; however, it allows playout delay decrease during silence periods only. Since the playout delay is increased during active speech, a speech extension algorithm is required. Therefore we have developed a novel, voicing-classification based speech extension algorithm for CELP speech coders. Though the complementary speech truncation algorithm is not needed for our adaptive playout mechanism, it is suitable for applications such as voice synchronization with other media.

关键词： linear predictive coding Jitter Delay effects Speech analysis Speech coding Decoding Speech synthesis Adaptive filters Digital signal processing Research and development

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TUF: tag-based unified fairness

TUF: tag-based unified fairness

引用

IEEE Annual Joint Conference: INFOCOM, IEEE Computer and Communications Societies

作者： A. Clerget W. Dabbous

ISBN: (纸本)0780370163

Finding an appropriate end-to-end congestion control scheme for each type of flow, such as real-time or multicast flows, may be difficult. But it becomes even more complex to have these schemes be friendly among themselves and with TCP. The assistance of routers within the network for fair bandwidth sharing among the flows is therefore helpful. However, most of the existing mechanisms that provide this fair sharing imply complex buffer management and maintaining flow state in the routers. We propose to realize this fair bandwidth sharing without per-flow state in the routers, using only a trivial queueing discipline. Packets are tagged near the source, depending on the nature of the flow. In the core of the network, routers use FIFO queues, and simply drop the packet with the highest tag value in case of congestion. Contrarily to other stateless fair queueing algorithms in the core routers, we do not try to maintain instantaneous flow rates equal. Instead, we take into account the responsiveness nature of the flows, and adjust loss rates such that average rates are equal. The novel approach of our scheme, called TUF, tag-based unified fairness, not only improves the overall fairness but enables us to maintain it in realistic environments, with non-negligible round trip times or bursty traffic, where other schemes fail. The corresponding cost is the need for models of the end to-end responsive natures of the flows.

关键词： Bandwidth Delay Protocols Communication system traffic control Traffic control Costs Internet linear predictive coding GSM Aggregates

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Candidate scheme for MP-MLQ search in G723.1

Candidate scheme for MP-MLQ search in G723.1

引用

IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC)

作者： Fu-Kun Chen Jar-Ferr Yang Department of Electrical Engineering National Cheng Kung University Tainan Taiwan

ISBN: (纸本)0780367200

For the purpose of low cost or low power consumption, a cost-effective solution with good trade-off between quality and complexity must be estimated for a small mobile device. We propose a condensed MP-MLQ algorithm to reduce the search computation required for G723.1. Simulation results show that the proposed method can reduce about 50/spl sim/82% the computational load for stochastic codebook search with perceptually intangible degradation. This reduction would be helpful to save power consumption and relax the CPU load for a mobile device in wireless communication, e.g., PALM, PDA.

关键词： Stochastic processes linear predictive coding Energy consumption Speech coding Speech analysis Distributed computing GSM Quantization Computational modeling Degradation

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Analysis of LSF switched-predictive vector quantisers

Analysis of LSF switched-predictive vector quantisers

引用

International Symposium on Signal Processing and Its Applications (ISSPA)

作者： R.C. de Lamare A. Alcaim CETUC Catholic University of Rio de Janeiro Rio de Janeiro Brazil

Switched-predictive vector quantisation (SPVQ) has proven to be more efficient than conventional predictive vector quantisation (PVQ) to reduce the bit rate of LSF parameters in speech coding applications. We perform a comparison between different SPVQ schemes and we propose an extension of the SPVQ technique which outperforms previously reported PVQ and SPVQ systems. The proposed method consists of 4 vector quantisers (VQ) which are switched on the basis of the best performance criterion.

关键词： Speech coding linear predictive coding Vector quantization Signal processing Bit rate Delay Signal processing algorithms Information filtering Information filters Nonlinear filters

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Speech recognition chip for monosyllables

Speech recognition chip for monosyllables

引用

Asia and South Pacific Design Automation Conference

作者： K. Nakamura Qiang Zhu S. Maruoka T. Horiyama S. Kimura K. Watanabe Graduate School of Information Science Nara Institute of Science and Technology Ikoma Nara Japan Fujitsu Laboratories Limited Japan Hitachi and Limited Japan

ISBN: (纸本)0780366336

In the paper, we present a real-time speech recognition chip for monosyllables such as A,B,..., etc. The chip recognizes up to 64 monosyllables based on the Hidden Markov Model (HMM), which is a well known speaker-independent recognition method. The chip accepts a short-speech frame including 256 16-bit digitized samples corresponding to 11.6 ms period, and outputs the 6-bit symbol code of monosyllables for 16 short-frames (corresponding to 185.6 ms). A learning circuit to update HMM parameters for the recognition chip has also been designed, and the recognition chip includes an interface to the learning circuit. We have fabricated the recognition chip by the VDEC Rohm 0.6 /spl mu/m process on a 4.5 mm/spl times/4.5 mm chip. We have also made a layout of the entire circuit including the learning circuit by the VDEC Rohm 0.35 /spl mu/m process on a 4.9 mm/spl times/4.9 mm chip.

关键词： Speech recognition Hidden Markov models linear predictive coding Speech analysis Vectors Circuits Algorithm design and analysis Hardware Feature extraction Speech coding

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