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检索条件"主题词=Linear Predictive coding"

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GLS-based TV-CAR speech analysis using forward and backward linear prediction

GLS-based TV-CAR speech analysis using forward and backward ...

引用

IEEE Workshop on Multimedia Signal Processing

作者： K. Funaki Computing and Networking Center University of Ryukyus Nakagami Okinawa Japan

We have already proposed novel robust parameter estimation algorithms of a time-varying complex AR (TV-CAR) model for analytic speech signals, which are based on GLS (generalized least squares) and ELS (extended least squares) and have shown that the methods can achieve robust speech spectrum estimation against additive white Gaussian. In these methods, forward prediction error is only used to calculate the MSE criterion. This paper proposes the improved TV-CAR speech analysis methods based on forward and backward linear prediction in which backward prediction error is also adopted to calculate the MSE criterion, viz., the MMSE and GLS-based algorithms using the forward and backward prediction. The experiments with natural speech and natural speech corrupted by white Gaussian demonstrate that the improved methods can achieve more accurate and more stable spectral estimation.

关键词： Speech analysis Signal analysis Natural languages Robustness linear predictive coding Speech processing Filters Least squares approximation Spectral analysis Least squares methods

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Sensitivity of MIL-STD-3005 MELP to packet loss on IP networks

Sensitivity of MIL-STD-3005 MELP to packet loss on IP networ...

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： E.J. Daniel K.A. Teague School of Electrical and Computer Engineering Oklahoma State University Stillwater OK USA

The sensitivity of military standard MELP (MIL-STD-3005) to randomly deleted frames on an IP network is studied. Frame deletion occurs when packets are lost or arrive too late to be useful in a voice-over-IP (VoIP) application. Network errors including packet loss, burst loss, jitter and out-of-order packets are simulated. The quality of reconstructed speech is measured by spectral distortion.

关键词： IP networks Distortion measurement Speech linear predictive coding Vocoders Quality of service Military standards Internet telephony Jitter Out of order

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Preprocessing of noisy speech for voice coders

Preprocessing of noisy speech for voice coders

引用

IEEE Workshop on Speech coding Proceedings

作者： T. Agarwal P. Kabal Department of Electrical & Computer Engineering McGill University Montreal QUE Canada

Accurate linear prediction coefficient (LPC) estimation is one of the key requirements for low bit-rate voice coding. Under harsh acoustic conditions, LPC estimation can become unreliable. This results in poor quality of encoded speech and introduces annoying artifacts. The paper presents a two-branch speech enhancement preprocessing scheme for low bit-rate voice coders. The scheme consists of two parallel denoising blocks. One block enhances the degraded speech for LPC estimation. Another block increases the perceptual quality of the speech to be coded. The goal of the paper is to design the two-branch scheme. Test results show that the two-branch scheme can provide better perceptual quality compared to conventional one-branch speech enhancement techniques in noisy environments.

关键词： linear predictive coding Speech enhancement Noise reduction Signal to noise ratio Working environment noise Speech processing Additive noise Acoustic noise Speech coding Acoustical engineering

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Noise suppression properties of synthesis error optimization

Noise suppression properties of synthesis error optimization

引用

IEEE Workshop on Speech coding Proceedings

作者： K. Lashkari T. Miki DoCoMo Communications Laboratory USA Inc. San Jose CA USA

Analysis-by-synthesis (AbS) speech codecs such as multi-pulse LPC and CELP compute the coefficients of the synthesis filter by minimizing the linear prediction (LP) error and the excitation function by minimizing the closed-loop synthesis error. In the presence of additive noise, the LPC coefficients will be affected by noise, resulting in a synthesized speech with degraded quality. This paper points out the noise advantages of computing both the filter parameters and the excitation function by minimizing the closed-loop synthesis error. By minimizing the synthesis error, the analysis and synthesis stages become more compatible. Furthermore, It is shown that unlike LP-based estimation, minimization of the synthesis error could lead to noise-free filter parameters for uncorrelated (colored) noise. An approach for reducing the synthesis error is presented and simulation results on both synthetic and real speech are given.

关键词： Speech synthesis linear predictive coding Additive noise Colored noise Speech analysis Speech codecs Nonlinear filters Speech enhancement Degradation Error analysis

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Spectral analysis of some typical Sino-Tibetan languages with special reference to North East Indian languages

Spectral analysis of some typical Sino-Tibetan languages wit...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： P.H. Talukdar S.K. Sarma University Science Instrumentation Center Assam India Indian Institute of Technology Guwahati Guwahati Assam India

We have made an attempt to study the spectral characteristics of two North East Indian languages, Assamese and Boro, coming from different genres. We have taken a few words with similar, partially similar, and dissimilar characteristics in their nature of utterance from Assamese and Boro. The spectral analysis revels that both the languages have a high degree of similarity at both voiced and unvoiced regions for words with the same meaning. LPC modeling of the random selection of similar, partially similar and dissimilar phonemes may not provide characteristic features of the two languages, but cepstral coefficient analysis of the LPC modeling may provide important information in studying the linguistic and phonetic distinction of the two languages and, finally, speaker identification and word verification.

关键词： Spectral analysis linear predictive coding Hidden Markov models Speech recognition Artificial neural networks Cepstral analysis Natural languages Speech analysis Instruments Training data

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Fast software implementation of MPEG advanced audio encoder

Fast software implementation of MPEG advanced audio encoder

引用

International Conference on Digital Signal Processing (DSP)

作者： I. Dimkoviae D. Milovanoviae Z. Bojkoviae PsyTEL Research Belgrade Serbia Faculty of Elcclrical Engineering Belgrade Serbia

An optimized software implementation of a high quality MPEG AAC-LC (low complexity) audio encoder is presented in this paper. The standard reference encoder is improved by utilizing several algorithmic optimizations (fast psycho-acoustic model, new tonality estimation, new time domain block switching, optimized quantizer and Huffman coder) and very careful code optimizations for PC CPU architectures with SIMD (single-instruction-multiple-data) instruction set. The psychoacoustic model used the MDCT filterbank for energy estimation and peak detection as a measure of tonality. Block size decision is based on local perceptual entropies as well as LPC analysis of the time signal. Algorithmic optimizations in the quantizer include loop control module modification and optimized Huffman search. Code optimization is based on parallel processing by replacing vector algebra and math junctions with their optimized equivalents with Intel/sup /spl reg// Signal Processing Library (SPL). The implemented codec outperforms consumer MP3 encoders at 30% less bitrate at the same time achieving encoding times several times faster than real-time.

关键词： Signal processing algorithms Psychoacoustic models Software quality Code standards Psychology Filter bank Energy measurement Entropy linear predictive coding Signal analysis

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Real-time speech-driven 3D face animation

Real-time speech-driven 3D face animation

引用

International Symposium on 3D Data Processing Visualization and Transmission

作者： Pengyu Hong Zhen Wen T.S. Huang Heung-Yeung Shum Coordinated Science Laboratory University of Illinois Urbana-Champaign USA Beckman Institute University of Illinois Urbana-Champaign USA Microsoft Research Laboratory-Beijing Beijing China

In this paper, we present an approach for real-time speech-driven 3D face animation using neural networks. We first analyze a 3D facial movement sequence of a talking subject and learn a quantitative representation of the facial deformations, called the 3D motion units (MUs). A 3D facial deformation can be approximated by a linear combination of the MUs weighted by the MU parameters (MUPs) - the visual features of the facial deformation. The facial movement sequence synchronizes with a audio track. The audio track is digitized and the audio features of each frame are calculated. A real-time audio-to-MUP mapping is constructed by training a set of neural networks using the calculated audio-visual features. The audio-visual features are divided into several groups based on the audio features. One neural network is trained per group to map the audio features to the corresponding MUPs. Given a new audio feature vector, we first classify it into one of the groups and select the corresponding neural network to map the audio feature vector to MUPs, which are used for face animation. The quantitative evaluation shows the effectiveness of the proposed approach.

关键词： Facial animation Neural networks Speech Shape linear predictive coding Vector quantization Context modeling Cepstral analysis Delay effects Visual databases

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An interface protocol component modeling language

An interface protocol component modeling language

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Annual IEEE International ASIC/SOC Conference

作者： Chang-Yul Yun Kyoung-Son Jhang Department of Computer Engineering ChungNam National University Taejon South Korea

ISBN: (纸本)0780374940

Reusing IPs requires designers to perform interface protocol related tasks such as writing test benches and designing interface protocol conversion circuits, e.g, wrappers for IPs. The results of those tasks usually include the interface protocol components for the corresponding IPs, similar to bus protocol components of the bus functional models. Interface protocols of most IPs can be abstracted in transactions. This paper presents a transaction-oriented interface protocol description language which models interface protocol components recognizing or executing transactions over the given interface ports. In addition, we describe a target structure of the synthesizable interface protocol component together with its application to an IP wrapper design. The proposed approach not only reduces re-works on the interface protocol components but also enables the methodology that can be called "transaction-based interface design or synthesis".

关键词： Protocols linear predictive coding Circuit synthesis Circuit testing Design engineering Writing Performance evaluation Master-slave Hardware design languages Computer interfaces

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On the relationship between root interpolation and LSP interpolation

On the relationship between root interpolation and LSP inter...

引用

Asilomar Conference on Signals, Systems & Computers

作者： K. Lashkari DoCoMo Communications Laboratory USA Inc. San Jose CA USA

Line spectrum pair (LSP) was first introduced by Itakura (1975) as an alternative representation for the LPC spectral information. It was later found that LSPs have very desirable quantization and interpolation properties. Interpolation of LSPs at the frame boundaries results in a synthesized speech that is smoother and has higher quality. This paper compares the two methods of interpolation: 1) interpolating the LSPs derived from the LPC coefficients vs. 2) computing the LSPs from interpolated roots of the LPC polynomial. Experimental results obtained using four to six second long male and female sentences suggest that these two sets of LSPs are effectively equivalent.

关键词： Interpolation Nonlinear filters Polynomials linear predictive coding Speech synthesis Quantization Dynamic range Codecs Laboratories Signal synthesis

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Spectral distortion in G.729 due to LPC quantization

Spectral distortion in G.729 due to LPC quantization

引用

IEEE (first ITW conference of the year) Information Theory Workshop (ITW)

作者： G. Radha Krishna M. Banerjee B.A. Vani Dept. Of Electrical Comm. Engg. Indian Institute of Technology Chennai India CAD Lab Indian Institute of Science Bangalore India

In the G.729 speech coder, linear predictive coding (LPC) parameters are converted into line spectral frequencies (LSF) and then quantized. Spectral distortion, due to quantization error at various stages is calculate... 详细信息

关键词： linear predictive coding Bandwidth Vector quantization Speech coding Table lookup Distortion measurement Digital signal processing Books Speech processing

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