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检索条件"主题词=Linear predictive coding"

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A new developed speech recognition education software in teaching LPC and LPC-derived cepstrum

A new developed speech recognition education software in tea...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Hong Kai Sze S.-H. Salleh A.Z. Sha'ameri Digital Signal Processing Group Faculty of Electrical Engineering Universiti Teknologi Malaysia Malaysia

Speech processing systems are highly complex and teaching students in this subject matter with the underlying technologies can be a challenging task. Thus education software is needed to provide the students with more convenient way to study, perform experiments and develop his/her first speech recognition application without an instructor. The Speech Recognition Virtual Classroom (SRVC) was developed as education software in teaching speech recognition. There are various modules, like signal analysis, dynamic time warping, vector quantization and hidden Markov models (HMM). These modules are integrated into a single environment that allows easy learning through experiments with real speech data.

关键词： Speech recognition Education linear predictive coding Cepstrum Hidden Markov models Speech processing Software performance Application software Signal analysis Vector quantization

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A text-independent speaker identification system using PARCOR and AR model

A text-independent speaker identification system using PARCO...

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Chia-Hsiung Liu O.T.-C. Chen Signal and Media Laboratories Department of Electrical Engineering National Chung Cheng University Chiayi Taiwan

In this work, we propose the partial-correlation (PARCOR) coefficients scheme to model the cross areas of the several cylinders from the vocal tract. By using the relationship of the acoustic impedance proportional to the reciprocal of cross areas, the ratios of cross areas between each neighboring cylinders are used to model a speaker's vocal tract. The autoregressive model (AR model) is performed on the speech residual signals, that are produced from the inverse vocal tract transform based on the PARCOR, to generate features. These features with the conventional features from the Mel-Frequency Cepstral Coefficient (MFCC) are used for the identification engine of the Gaussian Mixture Model (GMM). According to our computer analyses in the TIMIT speech database, the proposed system can yield better identification performance than the conventional approach.

关键词： Speech analysis Timing Engine cylinders Cepstral analysis Signal analysis Laboratories Impedance Signal generators linear predictive coding Speech recognition

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Split VQ and predictive split VQ of LSP parameters in packet networks

Split VQ and predictive split VQ of LSP parameters in packet...

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IEEE Digital Signal Processing Workshop

作者： F. Merazka D. Berkani Ecole Nationale Polytechnique (ENP) Algiers Algeria

ISBN: (纸本)0780381165

In many speech coding systems, the LPC coefficients are transformed to the Line Spectrum Pairs (LSP) parameters which are very effective representation for quantization of the LPC information. LSP representation consumes a large part of the total bit rate of the coder. Typically, the LSP are highly correlated from one frame to the next one, and a considerable reduction in bit rate can be achieved by exploiting this interframe correlation. However, interframe LSP coding can cause error propagation when frame erasures occur. In this paper, we compare the erasure performance of a predictive split vector quantizer to that of split vector quantizer based on the ITU G723.1 standard coder. Our results show that a 25 bits/frame split vector quantizer improves the average of spectral distortion of the 24 bits/frame predictive split vector quantizer based on The ITU G.723.1 for different loss rates.

关键词： Intelligent networks Vector quantization Speech coding linear predictive coding Bit rate Code standards Filters Robustness Source coding predictive models

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Multilayered and columnar competitive networks for spoken word recognition

Multilayered and columnar competitive networks for spoken wo...

引用

International Conference on Neural Information Processing

作者： S. Kurogi T. Sasaki M. Itakura T. Nishida Kyushu Institute of Technology Kitakyushu Japan

We have presented a multilayered and columnar competitive network involving competitive associative nets (CANs) and adaptive vector quantization nets (AVQNs) for spoken word recognition. Although the network has shown good performance in recognition rate, it requires a relatively large calculation time owing to the CANs. So, here, we present a new network replacing the CANs by a conventional feature extractor and an additional AVQNs, where as the feature extractor we use one of the three conventional methods: RLS (recursive least squares) for extracting LPCs, LSL (least squares lattice) for PARCOR coefficients, and normalized LSL for normalized PARCOR coefficients. As a result of experiments, the normalized LSL shows almost the same performance as the original network in recognition rate while reducing the calculation time.

关键词： Feature extraction Least squares methods linear predictive coding Data mining Vector quantization Signal processing Lattices Ear Function approximation Speech processing

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Modified very low bit rate coder for wireless messaging applications

Modified very low bit rate coder for wireless messaging appl...

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National Conference on Radio Science (NSRC)

作者： M.E.M. Nasr Department of Electronics & Cominun E J I. Faculty of Engineering Tanta University Egypt

This paper describes a modified variable coder with an average bit-rate of 1200 bps for wireless messaging including voice paging and voice e-mail delivery to a PC or a hand held device. The coder uses LPC based analysis/synthesis with Zinc function excitation for voiced frames, a plosive and pitch detector to achieve good synthetic voice quality at this rate. Further, a frame deemed unvoiced has the RMS value of its LPC residual quantized and sent to the decoder. The proposed system achieved an MOS of 3.1, while conventional MELP achieved an MOS of 3.3 in a subjective quality test. Further, the coder has been simulated on a workstation and a laptop PC running windows NT. Hence the coder can be used from MS Outlook to send and receive voice e-mails. The performance of the coder under random bit errors is also presented. It has been found that only at error rates of 10/sup -2/ and higher does the degradation becomes objectionable.

关键词： Bit rate Electronic mail linear predictive coding Paging strategies Speech analysis Zinc Detectors Decoding System testing Workstations

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A 2.4/1.2 kbps SB-LPC based speech coder: the Turkish NATO STANAG candidate

A 2.4/1.2 kbps SB-LPC based speech coder: the Turkish NATO S...

引用

IEEE Workshop on Speech coding Proceedings

作者： S. Villette K.T. Al Naimi C. Sturt A.M. Kondoz H. Palaz Centre for Communication Systems Research University of Surrey Guildford Surrey UK National Research Institute of Electronics and Cryptology Kocaeli Turkey

NATO military communications require high quality voice communications for their military missions. Currently the performance of in-place voice coding algorithms is unacceptable in the harsh tactical acoustic conditions where NATO commanders operate, such as tracked vehicles or helicopters. A new generation of 2.4/1.2 kbps speech coding algorithms has been developed which far exceed the quality of service of the existing speech coding algorithms currently used by NATO. This has led to a decision by NATO to standardize a new algorithm to suit its need, through a competition. A candidate coder based on the split-band LPC (SB-LPC) principle has been developed to fit the requirements of the competition, and has been entered by Turkey in the NATO STANAG competition. Its details are presented in this paper.

关键词： linear predictive coding Speech coding Background noise Military communication Quantization Noise reduction Speech enhancement Speech processing Filters Frequency domain analysis

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Constant delay and rate coding of speech spectral envelope at 11 bits/frame

Constant delay and rate coding of speech spectral envelope a...

引用

IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： E. Yakhnich Y. Bistritz Department of Electrical Engineering Tel-Aviv University Tel-Aviv Israel

Parametric vocoders are used to attain very low rate compression. A new method for segment coding of a speech spectral envelope is proposed, based on coding of constant length segments. It is a computationally efficient method for LPC (linear predictive coding) parameter coding based on TD (temporal decomposition). It attains average LSD (log-spectrum distortion) less than 1 dB at 11 bits/frame. Thus, it saves 56% of the LPC parameter bit budget which is 26% of the total NELP2400 vocoder bit budget. The method offers constant delay and relatively low complexity.

关键词： Delay Speech coding Quantization linear predictive coding Vectors Vocoders Bit rate Interpolation predictive models Frequency

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A methodology for dynamic power consumption estimation using VHDL descriptions

A methodology for dynamic power consumption estimation using...

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Symposium on Integrated Circuits and Systems Design (SBCCI)

作者： J.M.S. Alcantara A.C.C. Vieira F. Galvez-Dur V.C. Alves

Power consumption became an important feature to be considered in system implementations. This work presents a methodology for dynamic power consumption estimation using hardware descriptions written in VHDL; a library with information for transitions and power consumption for all components of the target library is created. A case study for the KASUMI cryptographic algorithm is reported. This algorithm was chosen to compose the 3rd Generation Partnership Project (3GPP) security functions for mobile systems. Restrictions imposed by the 3GPP to the hardware implementation of the KASUMI cryptographic algorithm were analyzed and satisfied; our dynamic power consumption estimation methodology is used. Only CMOS technologies are discussed in this paper.

关键词： Energy consumption CMOS technology Cryptography CMOS logic circuits CMOS digital integrated circuits Iron linear predictive coding Electrical capacitance tomography Information security Power system security

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Low temperature operation of ultra-thin gate oxide sub-0.1 /spl mu/m MOSFETs

Low temperature operation of ultra-thin gate oxide sub-0.1 /...

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European Workshop on Low Temperature Electronics

作者： B. Cretu F. Balestra G. Ghibaudo G. Guegan IMEPLPCS ENSERG Grenoble France CEA-LETI Grenoble France

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Dither-like data hiding in multistage vector quantization of MELP and G.729 speech coding

Dither-like data hiding in multistage vector quantization of...

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Asilomar Conference on Signals, Systems & Computers

作者： Pao-Chi Chang Hsin-Min Yu Department of Electrical Eng. National Central University Chungli Taiwan

ISBN: (纸本)0780375769

We present a speech data hiding technique that utilizes the characteristics of multistage vector quantization (MSVQ) and subtractive dithering to maintain high speech reconstruction quality. The last stage of MSVQ is used to store the data to be embedded. Similar to subtractive dithering, the noise-like hidden data is subtracted from the signal at the first stage of the encoder and added back at the MSVQ decoder. As a result, the degradation caused by hiding secret data is significantly reduced compared to the traditional simple substitution method.

关键词： Data encapsulation Vector quantization Speech coding Decoding Speech enhancement Degradation linear predictive coding Art Cryptography Protection

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