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检索条件"主题词=Linear predictive coding"

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Modelling energy flow in the vocal tract with applications to glottal closure and opening detection

Modelling energy flow in the vocal tract with applications t...

引用

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

作者： D.M. Brookes H.P. Loke Department of Electrical & Electronic Engineering Imperial College London London UK

The pitch-synchronous analysis that is used in several areas of speech processing often requires robust detection of the instants of glottal closure and opening. In this paper we derive expressions for the flow of acoustic energy in the lossless-tube model of the vocal tract and show how linear predictive analysis may be used to estimate the waveform of acoustic input power at the glottis. We demonstrate that this signal may be used to identify the instants of glottal closure and opening during voiced speech and contrast it with the LPC residual signal that previous authors have used for this purpose.

关键词： Lips Acoustic waves Signal processing linear predictive coding Speech analysis Acoustic reflection Power engineering and energy Educational institutions Speech processing Robustness

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A 4 kb/s toll quality harmonic excitation linear predictive speech coder

A 4 kb/s toll quality harmonic excitation linear predictive ...

引用

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

作者： S. Yeldener Lucent Technologies Bell Laboratories Innovations Clarksburg MD USA

The harmonic excitation linear predictive speech coder (HE-LPC) is a technique derived from MBE and MB-LPC type of speech coding algorithms. The HE-LPC coder has the potential of producing high quality speech at 4.8 kb/s and below. This coder employs a new pitch estimation and voicing technique. In addition, new DCT based LPC and residual amplitude quantization techniques have been developed. The 4 kb/s HE-LPC coder with a 14th order LPC filter was found to produce much better speech quality than the various low rate speech coding standards, including 3.6 kb/s INMARSAT Mini-M AMBE vocoder. During formal ITU ACR test, the 4 kb/s HE-LPC vocoder was found to produced equivalent performance to 32 kb/s ADPCM and G.729 for both flat and modified IRS filtered clean input speech conditions. The HE-LPC algorithm can also be extended to cover bit rates between 1.2 and 8 kb/s range depending on the application.

关键词： Vocoders Speech coding linear predictive coding Bit rate Power harmonic filters Cutoff frequency Quantization Speech codecs Frequency domain analysis Ear

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MELP vocoder using lifting wavelet transform

MELP vocoder using lifting wavelet transform

引用

IEEE Pacific Rim Conference on Communications, Computers and Signal Processing

作者： N. Aoki K. Takaya Research Institute for Electronic Science Hokkaido University Sapporo Japan Telecommunications Research Laboratory Saskatoon SAS Canada Department Electrical Engineering University of Saskatchewan Saskatoon SAS Canada

A MELP (mixed excitation linear prediction) vocoder produces much improved voice quality compared with the traditional LPC vocoder. An LPC synthesis filter which mimics the vocal tract characteristics is excited with a mixture of pulse and noise instead of using either of them. This paper describes an implementation of the MELP vocoder which uses the lifting wavelet transform in place of the band-pass filter bank required for subband division in the synthesis and analysis of the MELP vocoder. A new method to generate an appropriate glottal waveform is also described. In addition, three kinds of fluctuations observed in the steady parts of voiced speech are incorporated to enhance the naturalness of synthesized speech. This MELP vocoder was realized on a DSP evaluation module to confirm the feasibility of real-time implementation.

关键词： Vocoders Wavelet transforms Speech synthesis linear predictive coding Band pass filters Speech enhancement Filter bank Wavelet analysis Fluctuations Digital signal processing

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Buried oxide fringing capacitance: a new physical model and its implication on SOI device scaling and architecture

Buried oxide fringing capacitance: a new physical model and ...

引用

IEEE SOI-3D-Subthreshold Microelectronics Technology Unified Conference (S3S)

作者： T. Ernst S. Cristoloveanu STMicroelectronics Crolles France ENSERG LPCS Grenoble France

Fringing fields into the buried oxide and substrate depletion region stand as a key limiting factor for SOI MOSFET channel length reduction beyond 0.1 /spl mu/m. In fully-depleted (FD) SOI transistors, they cause a strong DIBL enhancement and a parasitic back channel conduction. On the other hand, in partially-depleted (PD) devices, the back channel control is even more difficult. The understanding and modeling of this phenomenon is of major interest, especially for RF SOI applications on high resistivity substrates where the depleted substrates behave as dielectrics. Various solutions to reduce these drawbacks are envisaged, such as buried oxide shrinking or double gate devices (Colinge, 1997; Cristoloveanu and Li, 1995). Thus far, the fringing field effect was ignored or merely included in FD analytical models by use of adjustable parameters. This paper presents a simple physical model for the evaluation of short channel effects induced by the BOX and substrate depletion. We analyze the lateral drain field penetration in the BOX and substrate, and calculate the related fringing capacitances. The model serves to anticipate the buried oxide scaling and substrate resistivity effects and to suggest the "ground plane" (GP) concept (Ernst and Cristoloveanu, 1999; Wong et al, 1998) as a suitable architecture for deep sub-micron SOI MOSFETs.

关键词： Capacitance Conductivity Dielectric substrates Silicon Semiconductor films linear predictive coding Microelectronics MOSFET circuits Radio frequency Analytical models

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Speech-pitch detection using maximum likelihood algorithm

Speech-pitch detection using maximum likelihood algorithm

引用

Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： N.M. Botros R.S. Adamjee Department of Electrical Engineering Southem Illinois University Carbondale Carbondale IL USA

Presents a robust algorithm for pitch detection and estimation. The algorithm is based on implementation of the maximum-likelihood estimation. The speech signal is digitized with 8 kHz, 10-bit Analog-to-Digital Converter (ADC) and the digitized data are stored in a PC where it is segmented into frames each of 32 msec. A theoretical analysis has been conducted in this paper to find the maximum-likelihood epoch determination signal that represent the Glottal Closure Instant (GCI) from which the period can be determined. The Hilbert transform is incorporated into the algorithm to improve the overall performance of the algorithm. Utterances from three male and three female speakers have been used to test the algorithm. The algorithm has performed reliably and detected correctly voiced, unvoiced, and mixed modes of speech.

关键词： Maximum likelihood detection Maximum likelihood estimation Speech analysis Robustness Testing linear predictive coding Probability density function Gaussian processes Computer displays Analog-digital conversion

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Speech restoration from LPC parameters corrupted by channel noise

Speech restoration from LPC parameters corrupted by channel ...

引用

IEEE Region 10 International Conference TENCON

作者： A.B. Premkumar A. Eeluang A.S. Madhukumar School of Applied Science Nanyang Technological University Singapore

Restoration of speech parameters that have been degraded by noisy channel is of paramount importance, since mutilated speech parameters such as linear predictive coefficients (LPC) may render speech totally irrecoverable. The paper provides a technique using discrete dyadic wavelet transforms as a preprocessing stage prior to LPC extraction. We show that this step makes the speech parameters more robust to noise perturbation in the channel and hence makes recovery of speech possible. We discuss the effects of applying a single level and a three level transformation to speech prior to compression and compare the quality of recovered speech in both cases. We also apply a Wiener filter to the wavelet coefficients and show that the recovered speech is of appreciable quality.

关键词： Speech enhancement linear predictive coding Additive noise Degradation Speech synthesis Discrete wavelet transforms Speech processing Additive white noise Noise reduction Signal synthesis

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A perturbation-based pre-processing algorithm for CELP-coders

A perturbation-based pre-processing algorithm for CELP-coder...

引用

IEEE Workshop on Speech coding Proceedings

作者： J. Jensen S.H. Jensen E. Hansen Center for PersonKommunikation (CPK) University of Aalborg Denmark

A novel pre-processing algorithm for CELP-coders is proposed. The algorithm aims at perturbing the original signal slightly, such that the perturbed signal is subjectively indistinguishable from the original but can be coded more effectively. A key feature of the algorithm is the possibility of controlling the frequency domain properties of the perturbations. Preliminary simulations with the proposed algorithm in combination with a CELP-like coder indicate improvements in terms of segmental SNR and subjective speech quality.

关键词： Decoding Equations Speech coding Signal synthesis Filters Frequency domain analysis Distortion Encoding linear predictive coding

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Low bit rate line spectrum pair (LSP) quantizer with application to speech coding

Low bit rate line spectrum pair (LSP) quantizer with applica...

引用

National Conference on Radio Science (NSRC)

作者： R. El-Kouatly S. El-Ramly Elect. and Comm. Engineering Department Faculty of Engineering Ain Shams University Egypt

This paper proposes a new efficient line spectrum pair (LSP) quantizer, which is based on the combination of the difference LSP techniques and moving average (MA) interframe prediction. The difference technique is used to remove the correlation between adjacent LSP coefficients in the present frame, while the moving average (MA) prediction removes the correlation between the corresponding coefficients in successive frames. Two techniques are used for residual quantization, the first is a low complexity scalar quantizer with 20-25 bits per frame with 20 ms frame size. The second is of higher complexity but lower bit rate which uses 14 bits per frame two stages vector quantizer (VQ). The DMAP quantization technique yields a good performance comparable to the moving average two-stage split vector quantizer used in the ITU-T G.729 standard.

关键词： Bit rate Speech coding Quantization linear predictive coding predictive models Vectors Pulse modulation Information filtering Information filters Nonlinear filters

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A comparative study of a hidden Markov model detector for atrial fibrillation

A comparative study of a hidden Markov model detector for at...

引用

IEEE Workshop on Neural Networks for Signal Processing

作者： B. Young D. Brodnick R. Spaulding GE Marquette Medical Systems Milwaukee WI USA

A comparative study of several atrial fibrillation (AF) detection algorithms was done to determine the algorithm best suited for use in real clinical environments to detect AF in ambulatory ECGs. The algorithms that were investigated for this paper are based on the Hidden Markov Model (HMM), measures of variance, linear predictive coding, and measurement of approximate entropy (AE). Based on the results from the test data set, the HMM algorithm performed best for this application. In general, there is little difference between the performance of the HMM and AE algorithms. However, the implementation of the HMM algorithm is more computationally efficient. Because of the large amount of data that must be analyzed in ambulatory ECG recordings, the computational efficiency must be considered as an issue of practicality. Review of the data illuminated some of the strengths and weaknesses of the various algorithms. Variance measures performed with either high sensitivity or high positive predictivity, but were not able to achieve a desirable operating point that had both acceptable sensitivity and positive predictivity. Although AE and LPC had acceptable sensitivity and positive predictivity, the HMM performed even better than both of these in terms of overall error rate. It would seem that an observational model such as the HMM, fits the data better than parametric models such as AE and LPC. Finally, as the computing power of medical systems increases, more sophisticated algorithms may be exploited in ways that leads to more accurate computerized ECG interpretation.

关键词： Hidden Markov models Detectors Electrocardiography linear predictive coding Performance evaluation Atrial fibrillation Detection algorithms Entropy Testing Computational efficiency

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An adaptive post-filtering technique based on a least squares approach

An adaptive post-filtering technique based on a least square...

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IEEE Workshop on Speech coding Proceedings

作者： A. Mustapha S. Yeldener COMSAT Laboratories Clarksburg MD USA

This paper presents an adaptive time-domain post-filtering technique based on the least squares approach and modified Yule-Walker (MYW) filter. Conventionally, post-filtering is derived from an original LPC spectrum. In general, this time-domain technique produces unpredictable spectral tilt that is hard to control by the modified LPC synthesis, inverse and high pass filtering and hence introduces muffling in the speech quality. Other approaches of designing post-filters were developed in the frequency domain which can only be used in sinusoidal based speech coders. We have also developed a new time-domain post-filtering technique which eliminates the problem of spectral tilt in the speech spectrum and can be applied to various speech coders. The new post-filter has a flat frequency response at the formant peaks of the speech spectrum. This post-filtering technique has been used in a 4 kb/s harmonic excitation linear predictive coder (HE-LPC) and subjective listening tests have indicated that this technique outperforms the conventional one in both one and two tandem connections.

关键词： Least squares methods Time domain analysis linear predictive coding Speech synthesis Adaptive filters Filtering Frequency domain analysis Frequency response Power harmonic filters Testing

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