检索结果-内蒙古大学图书馆

On predictive coding for Erasure Channels Using a Kalman Framework

IEEE TRANSACTIONS ON SIGNAL PROCESSING 2009年第11期57卷 4456-4466页

作者： Arildsen, Thomas Murthi, Manohar N. Andersen, Soren Vang Jensen, Soren Holdt Aalborg Univ Dept Elect Syst DK-9220 Aalborg Denmark Univ Miami Coral Gables FL 33146 USA

We present a new design method for robust low-delay coding of autoregressive (AR) sources for transmission across erasure channels. It is a fundamental rethinking of existing concepts. It considers the encoder a mechanism that produces signal measurements from which the decoder estimates the original signal. The method is based on linear predictive coding and Kalman estimation at the decoder. We employ a novel encoder state-space representation with a linear quantization noise model. The encoder is represented by the Kalman measurement at the decoder. The presented method designs the encoder and decoder offline through an iterative algorithm based on closed-form minimization of the trace of the decoder state error covariance. The design method is shown to provide considerable performance gains, when the transmitted quantized prediction errors are subject to loss, in terms of signal-to-noise ratio (SNR) compared to the same coding framework optimized for no loss. The design method applies to stationary auto-regressive sources of any order. We demonstrate the method in a framework based on a generalized differential pulse code modulation (DPCM) encoder. The presented principles can be applied to more complicated coding systems that incorporate predictive coding as well.

关键词： Differential pulse code modulation (DPCM) erasure channels joint source-channel coding Kalman filtering linear predictive coding quantization

来源：评论

学校读者我要写书评

暂无评论

Self-supervised Prosody Learning at Phoneme-level with Momentum Contrast for Speech Synthesis

Self-supervised Prosody Learning at Phoneme-level with Momen...

引用

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

作者： Zhao-Ci Liu Ya-Jun Hu Liping Chen Zhen-Hua Ling National Engineering Research Center of Speech and Language Information Processing University of Science and Technology of China Hefei P.R.China iFLYTEK Research iFLYTEK Co. Ltd. China

ISBN: (数字)9798350368741

ISBN: (纸本)9798350368758

This paper investigates leveraging large-scale speech data to enhance prosodic modeling in speech synthesis, and introduces a model named SP2MC which achieves self-supervised prosody learning at phoneme-level with momentum contrast. This model incorporates dual convolutional encoders for speech and linear predictive coding (LPC) residual inputs to generate phoneme-level embeddings, which are masked and processed by a Transformer model to produce prosody representations. Two supervision modules are employed to generate phoneme-level supervision from speech waveforms and residuals. Momentum contrast is utilized to manage negative sample selection in contrastive learning. Finally, the SP2MC representations are integrated into a Fastspeech2-based acoustic model for speech synthesis. Experimental results indicate that the naturalness of speech synthesized by the proposed method is significantly better than that of baselines.

关键词： Convolutional codes Speech coding Convolution predictive models Speech enhancement Transformers Acoustics Data models linear predictive coding Text to speech

来源：评论

学校读者我要写书评

暂无评论

A neural network model for nonlinear predictive coding in Fock space

A neural network model for nonlinear predictive coding in Fo...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： R.J.P. deFigueiredo Department of Electrical and Computer Engineering and Department of Mathematics University of California Irvine CA USA

A generic nonlinear autoregressive (AR) model for a random time series is presented. The model is obtained by a nonlinear predictive coding (NLPC) approach which expresses the minimum mean square error estimate of the current value of the series as a Volterra series in terms of its immediate N preceding values. This Volterra series is assumed to belong to a generalized Fock Hilbert space F. In the second stage, which is parametric, the model parameters, which are coefficients of a linear combination of known nonlinear random functions of the data, are obtained by linear mean square estimation. The implementations of the model and of the estimator appear respectively as two layer recurrent and feedforward neural networks.< >

关键词： Neural networks predictive models predictive coding Intelligent networks linear predictive coding Equations Mathematics Mathematical model Mean square error methods Hilbert space

来源：评论

学校读者我要写书评

暂无评论

I Predict, Therefore I Am: Perturbed predictive coding Under Ketamine and in Schizophrenia

引用

BIOLOGICAL PSYCHIATRY 2017年第6期81卷 465-466页

作者： Corlett, Philip R. Yale Univ Dept Psychiat New Haven CT 06511 USA

来源：评论

学校读者我要写书评

暂无评论

predictive fusion coding of spatio-temporally correlated sources

Predictive fusion coding of spatio-temporally correlated sou...

引用

33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Ramaswamy, Sharad Rose, Kenneth Univ Calif Santa Barbara Dept ECE Santa Barbara CA 93106 USA

ISBN: (纸本)9781424414833

This paper considers the problem of predictive fusion coding for storage of multiple spatio-temporally correlated sources so as to enable efficient selective retrieval of data from subsets of sources as designated by future queries. Only statistical information about future queries is available during encoding. While temporal correlations can be exploited by coding over large blocks, the growth in encoding complexity renders this approach impractical and hence the interest in a low complexity predictive coding approach. However, the design of optimal predictive fusion coding systems is considerably complicated by the presence of the prediction loop, and the potentially exponential growth of the query sets. We propose a complexity-constrained predictive fusion coder and derive an iterative algorithm for its design, which is based on the "Asymptotic Closed Loop" framework and hence, circumvents convergence and stability issues of traditional predictive quantizer design. The proposed predictive fusion coder optimizes the distortion - retrieval rate tradeoff, given a fixed storage capacity, and provides significant gains over storage schemes that perform only joint compression or memoryless fusion coding of all sources.

关键词： multisensor systems database query processing linear predictive coding vector quantization data compression

来源：评论

学校读者我要写书评

暂无评论

Systolic Codebook Generation

引用

IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1993年第2期1卷 144-149页

作者： Beresford-Smith, Bryan Breckling, Jens Schroeder, Heiko Summons, Peter Australian Natl Univ Comp Sci Lab Canberra ACT 2601 Australia Univ Newcastle Callaghan NSW 2308 Australia

To tailor speech recognition systems for different speakers, fast codebook generation is essential. VLSI implementations of codebook generators can help to increase their speed of generation. Two systolic codebook generators are presented which are based on finding clusters in a multidimensional space with a suitable dissimilarity measure. They are implemented on two novel VLSI architectures. For segments of speech represented by a sequence of k LPC vectors, the first version finds r large clusters in time O(rk) and the second version finds the maximal r clusters in time O( r log k).

关键词： Speech recognition linear predictive coding Australia Very large scale integration Speech processing Vector quantization Iterative algorithms Multidimensional systems Extraterrestrial measurements Hidden Markov models

来源：评论

学校读者我要写书评

暂无评论

Comparison of speech parameterization techniques for the classification of speech disfluencies

引用

TURKISH JOURNAL OF ELECTRICAL ENGINEERING AND COMPUTER SCIENCES 2013年第Sup1期21卷 1983-1994页

作者： Fook, Chong Yen Muthusamy, Hariharan Chee, Lim Sin Bin Yaacob, Sazali Bin Adom, Abdul Hamid Univ Malaysia Perlis Sch Mechatron Engn Arau Perlis Malaysia

Stuttering assessment through the manual classification of speech disfluencies is subjective, inconsistent, time-consuming, and prone to error. The aim of this paper is to compare the effectiveness of the 3 speech feature extraction methods, mel-frequency cepstral coefficients, linear predictive coding (LPC)-based cepstral parameters, and perceptual linear predictive (PLP) analysis, for classifying 2 types of speech disfluencies, repetition and prolongation, from recorded disfluent speech samples. Three different classifiers, the k-nearest neighbor classifier, linear discriminant analysis-based classifier, and support vector machine, are employed for the classification of speech disfluencies. Speech samples are taken from the University College London Archive of Stuttered Speech and stuttered events are identified through manual segmentation. A 10-fold cross-validation method is used for testing the reliability of the classifier results. The effect of the 2 parameters (LPC order and frame length) in the LPC- and PLP-based methods on the classification results is also investigated. The experimental results reveal that the proposed method can be used to help speech language pathologists in classifying speech disfluencies.

关键词： Disfluent speech mel-frequency cepstral coefficient linear predictive coding perceptual linear predictive analysis support vector machine

来源：评论

学校读者我要写书评

暂无评论

ADVANCES IN SPEECH AND AUDIO COMPRESSION

引用

PROCEEDINGS OF THE IEEE 1994年第6期82卷 900-918页

作者： GERSHO, A Dept. of Econ. California Univ. Santa Barbara CA

Speech and audio compression has advanced rapidly in recent years spurred on by cost-effective digital technology and diverse commercial applications. Recent activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding. These algorithms exploit models of speech production and auditory perception and offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques for rates in the range of 4 to 16 kb/s. Techniques have also been emerging in recent years that offer enhanced quality in the neighborhood of 2.4 kb/s over traditional vocoder methods. Wideband audio compression is generally aimed at a quality that is nearly indistinguishable from consumer compact-disc audio. Subband and transform coding methods combined with sophisticated perceptual coding techniques dominate in this arena with nearly transparent quality achieved at bit rates in the neighborhood of 128 kb/s per channel.

关键词： 128 kbit/s 2.4 kbit/s 4 to 16 kbit/s audio compression audio signals auditory perception models bit rates code-excited linear prediction coding data compression digital technology encoding linear predictive coding perceptual coding techniques quality speech coding speech compression speech production models subband coding methods transform coding methods

来源：评论

学校读者我要写书评

暂无评论

A HUFFMAN-TYPE CODE GENERATOR WITH ORDER-N COMPLEXITY

引用

IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1990年第9期38卷 1619-1626页

作者： LU, MI CHEN, CF TATUNG INST TECHNOL DEPT ELECT ENGN TAIPEI TAIWAN

A new method with order-N complexity is presented to construct Huffman-type minimum-redundancy codes for N distinguished symbols in the source alphabets. This method includes a contraction process as well as an expansion process. The contraction process has (N-3) contraction stages. To reduce the data-transfer operations, (N-5) vacancies are reserved beforehand in the array which is used to store the probabilities of symbols. At each contraction stage, the number of the symbols whose probabilities are equal to or less than the sum of the two least probabilities is stored and named as the expansion index. Under the contraction process, the code lengths of the original symbols and those at each contraction stage increase monotonically. Since the proposed method gives a monotonically increasing code both in code length and in code value, its implementation becomes easy in VLSI technology, in microprocessor-based systems, or in software programming.< >

关键词： linear predictive coding Probability distribution Image coding Very large scale integration Software systems System testing Councils Data analysis Data structures Encoding

来源：评论

学校读者我要写书评

暂无评论

THEORETICAL AND EXPERIMENTAL RATE DISTORTION PERFORMANCE IN COMPRESSION OF AMBULATORY ECGS

引用

IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1991年第3期38卷 260-266页

作者： HAMILTON, PS TOMPKINS, WJ UNIV WISCONSIN DEPT ELECT ENGN MADISON WI 53706 USA

We compare ECG data compression algorithms based on signal entropy for a given mean-square-error (MSE) compression distortion. By defining the distortion in terms of the MSE and assuming the ECG signal to be a Gaussian process we are able to estimate theoretical rate distortion bounds from average ECG power spectra. These rate distortion bounds give estimates of the minimum bits per second (bps) required for storage of ECG data with a given MSE regardless of compression method. From average power spectra of the MIT/BIH arrhythmia database we have estimated rate distortion bounds for ambulatory ECG data, both before and after average beat subtraction. These rate distortion estimates indicate that, regardless of distortion, average beat subtraction reduces the theoretical minimum data rate required for ECG storage by approximately 100 bits per second (bps). Our estimates also indicate that practical ambulatory recording requires a compression distortion on the order of 11-mu-V rms. We have compared the performance of common ECG compression algorithms on data from the MIT/BIH database. We sampled and quantized the data to give distortion levels of 2, 5, 8, 11, and 14-mu-V rms. These results indicate that, when sample rates and quantization levels are chosen for optimal rate distortion performance, minimum data rates can be achieved by average beat subtraction followed by first differencing of the residual signal. Achievable data rates approximate our theoretical estimates at low distortion levels and are within 60 bps at higher distortion levels.

关键词： Rate-distortion Electrocardiography Databases Compression algorithms Data compression Entropy Estimation theory Monitoring Solid state circuits linear predictive coding

来源：评论

学校读者我要写书评

暂无评论

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案：

请选择收藏分类：

通借通还

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：