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LSP analysis and processing for speech coders

ELECTRONICS LETTERS 1997年第9期33卷 743-744页

作者： McLoughlin, IV Chance, RJ School of Electronic and Electrical Engineering The University of Birmingham Birmingham United Kingdom

linear prediction parameters within CELP coders are commonly represented by line spectral pairs (LSP), giving stable filters and efficient coding. However, LSP manipulation can also alter the frequencies of the represented signals. The authors use computationally efficient LSP manipulation to enhance the intelligibility of speech degraded by acoustic interference.

关键词： speech coding linear predictive coding

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The 1974 origins of VoIP

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IEEE SIGNAL PROCESSING MAGAZINE 2005年第4期22卷 87-90页

作者： Gray, RM Massachusetts Institute of Technology University of Southern California

This paper brings light on the digital signal processing (DSP) roots of a modern concept, voice over IP (VoIP). An example is also provided in which developments in DSP - speech coding, in particular - had a profound impact on the early development of the ARPANET, the ancestor of the Internet. The author shows how packet speech, recently rediscovered and made popular as VoIP, was first successfully demonstrated in 1974 on the ARPANET and how the Internet protocol (IP) emerged largely as a result of that effort.

关键词： Internet telephony Digital signal processing History Speech coding linear predictive coding Speech processing Image coding Collaboration ARPANET predictive models

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ZERO-BRANCH TREE ENcoding OF SPEECH PITCH VOICING SIGNALS

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IEEE TRANSACTIONS ON COMMUNICATIONS 1989年第11期37卷 1236-1239页

作者： GONCHAROFF, V BUKIET, B HAYNER, D University of Illinois Chicago IL USA

A computationally efficient, although suboptimal, tree encoding method for the pitch and voicing parameters of an LPC (linear predictive coding) vocoder is presented. It is shown that when pitch and voicing signals are combined, it becomes difficult to take advantage of linear predictors and at the same time avoid any errors in voicing. The modified pitch tree coder presented here solves this problem by incorporating a branch leading to zero pitch value from each node of the tree. Only two bits per analysis frame are used to convey the combined pitch/voicing signal, with 1.585 bits being used to encode nonzero pitch values.< >

关键词： Encoding Speech Jitter linear predictive coding Synchronization Frequency Bit rate Delay Clocks Vocoders

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A speech coding algorithm based on predictive coding

A speech coding algorithm based on predictive coding

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Data Compression Conference (DCC)

作者： S. Kwong K.F. Man Department of Computer Science City University of Hong Kong Hong Kong China

Summary form only given. A compression algorithm for high quality speech signal using predictive coding techniques is developed. Code-excited linear predictive coding (CELPC) is one of the key techniques to compress speech signal to a bit-rate around 4.8 Kbps. However, due to the heavy computational requirement in the CELPC and speech signals usually can be divided into two portions: namely the based-band and the high-band frequency range. A hybrid CELPC and voice excited linear predictive coding (VELPC) scheme is developed for speech coding to reduce the complexity of the original CELPC. In the algorithm, a speech signal is firstly divided into two portions, the based-band and high-band respectively, in frequency domain, and then the low portion is coded with CELPC and the high-band portion is coded with VELPC. The test experiments showed this new coder can produce synthesized speech with good quality at a better bit rates than the original CELPC. When using the coding methods for the base-band and the high-band signal, we must decide how to divide the speech signal into two portions. In choosing the bandwidth of the base-band signal, there is a trade-off between the coding quality and the bit rate. In our experiment, the bandwidth of the base-band signal is chosen as one fourth of that of the original speech. Subjective evaluation experiments were conducted to test the performance of the hybrid CELPC and VELPC technique. For speech signal sampled at 8 kHz, a bit rate of 4.0 kbps can be achieved with frame intervals of 23 ms. The experimental results showed that the quality of the synthesized speech using hybrid coding technique at the bit rate of 4.0 kbps was almost the same as that of the CELPC at the bit rate of 4.8 kbps.

关键词： Speech coding Prediction algorithms predictive coding Bit rate Speech synthesis linear predictive coding Testing Signal synthesis Bandwidth Compression algorithms

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A FREQUENCY-RATIO-BASED 12-BIT MOS PRECISION BINARY CURRENT SOURCE

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IEEE JOURNAL OF SOLID-STATE CIRCUITS 1984年第6期19卷 1029-1037页

作者： KWONG, EC BALDWIN, GL HORNAK, T Hewlett Packard Laboratories Palo Alto CA USA

A binary-weighted current generator that uses a weighted repetition-rate current-switching scheme is described. This scheme, as demonstrated in a monolithic MOS implementation, is rendered free from the disadvantages of a passive-resistive ladder network and is suitable for parallel-type digital-to-analog converter systems. linearity and precision in excess of 12 bits, with no postprocessing adjustments, are demonstrated.

关键词： Frequency Solid state circuits Signal processing Autocorrelation linear predictive coding Speech processing Adaptive signal processing Switching circuits Filter bank Speech analysis

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Prediction of Five-Year Survival Rate for Rectal Cancer Using Markov Models of Convolutional Features of RhoB Expression on Tissue Microarray

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IEEE-ACM TRANSACTIONS ON COMPUTATIONAL BIOLOGY AND BIOINFORMATICS 2023年第5期20卷 3195-3204页

作者： Pham, Tuan D. Prince Mohammad Bin Fahd Univ Ctr Artificial Intelligence Khobar 31952 Saudi Arabia

The ability to predict survival in cancer is clinically important because the finding can help patients and physicians make optimal treatment decisions. Artificial intelligence in the context of deep learning has been increasingly realized by the informatics-oriented medical community as a powerful machine-learning technology for cancer research, diagnosis, prediction, and treatment. This paper presents the combination of deep learning, data coding, and probabilistic modeling for predicting five-year survival in a cohort of patients with rectal cancer using images of RhoB expression on biopsies. Using about one-third of the patients' data for testing, the proposed approach achieved 90% prediction accuracy, which is much higher than the direct use of the best pretrained convolutional neural network (70%) and the best coupling of a pretrained model and support vector machines (70%).

关键词： Rectal cancer RhoB protein deep learning convolutional neural networks Markov models linear predictive coding vector quantization

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Perceptual coding of digital audio

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PROCEEDINGS OF THE IEEE 2000年第4期88卷 451-513页

作者： Painter, T Spanias, A Arizona State Univ Ctr Telecommun Res Dept Elect Engn Tempe AZ 85287 USA

During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low rest. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, consider-able research has been devoted to the development of algorithms for perceptually transparent coding of high,fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have non become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model I discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linens prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby(

关键词： AC-2 AC-3 advanced audio coding (AAC) MPEG ATRAC audio coding audio coding standards audio signal processing data compression digital audio radio (DAR) digital broadcast audio (DBA) filter banks high-definition TV (HDTV) linear predictive coding lossy compression modified discrete cosine transform (MDCT) MP3 MPEG MPEG-1 MPEG-2 MPEG-4 MPEG audio multimedia signal processing perceptual audio coding (PAC) perceptual coding perceptual model pseudoquadrature mirror filter (PQMF) psychoacoustic model psychoacoustics SDDS signal compression signal-processing applications sinusoidal coding subband coding transform coding

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Multichannel linear prediction method compliant with the MPEG-4 ALS

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IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES 2008年第3期E91A卷 756-762页

作者： Kamamoto, Yutaka Harada, Noboru Moriya, Takehiro NTT Corp NTT Commun Sci Labs Atsugi Kanagawa 2430198 Japan

A new linear prediction analysis method for multichannel signals was devised. with the goal of enhancing the compression performance of the MPEG-4 Audio Lossless coding (ALS) compliant encoder and decoder. The multichannel coding tool for this standard carries out an adaptively weighted subtraction of the residual signals of the coding channel from those of the reference channel, both of which are produced by independent linear prediction. Our linear prediction method tries to directly minimize the amplitude of the predicted residual signal after subtraction of the signals of the coding channel, and the method has been implemented in the MPEG-4 ALS codec software. The results of a comprehensive evaluation show that this method reduces the size of a compressed file. The maximum improvement of the compression ratio is 14.6% which is achieved at the cost of a small increase in computational complexity at the encoder and without increase in decoding time. This is a practical method because the compressed bitstream remains compliant with the MPEG-4 ALS standard.

关键词： linear predictive coding multichannel signal lossless compression and MPEG-4 ALS

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Autonomous control systems: Monitoring, diagnosis and tuning

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IEEE TRANSACTIONS ON SYSTEMS MAN AND CYBERNETICS PART A-SYSTEMS AND HUMANS 1996年第5期26卷 646-655页

作者： Doraiswami, R Stevenson, M Diduch, CP Department of Electrical Engineering University of New Brunswick Fredericton NB Canada University of New Brunswick Fredericton NB CA Dept. of Electr. Eng. New Brunswick Univ. Fredericton NB Canada

A systematic and unified approach which accomplishes performance monitoring, performance improvement and fault prediction in control systems is proposed. The feature vector which is a vector formed of the coefficients of the estimate of the sensitivity function and the influence matrix which is the Jacobian of the feature vector with respect to the physical parameter are shown to contain the relevant information to realize an autonomous control system. The feature vector is estimated using a robust, accurate and reliable linear predictive coding Algorithm (LPCA). The influence matrix is computed by perturbing the physical parameters one at a time and estimating the feature vectors for each case. The proposed scheme is evaluated both on simulated as well as on actual control systems.

关键词： Control systems Monitoring Jacobian matrices Vectors Robustness linear predictive coding Prediction algorithms Physics computing Computational modeling Control system synthesis

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JNDS IN THE LPC POLES OF SPEECH AND THEIR APPLICATION TO QUANTIZATION OF THE LPC FILTER

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第2期39卷 308-318页

作者： ERELL, A ORGAD, Y GOLDSTEIN, JL SRI INT SPEECH RES LABMENLO PKCA 94025

The poles of a linear prediction coder (LPC) are directly related to speech formants;thus, quantizing the poles in an LPC vocoder is the most direct method for incorporating knowledge on formant perception in the vocoder design. For the purpose of guiding a pole quantization scheme, a psychophysical experiment was performed to measure just noticeable differences (JND's) in the frequency and radius of the poles. The results can be associated with formant JND's, extending presently existing data. The frequency JND's, measured up to a formant frequency of 4 kHz, are quantified as distributions with means that are increasing functions of formant frequency and band-width. An example of a pole quantization scheme, based on the JND data, is presented and found to be significantly superior to common scalar quantization methods of the LPC-PARCOR coefficients. The pole quantization scheme is also found to be almost comparable, both in quality and bit consumption, to vector quantization.

关键词： linear predictive coding Quantization Vocoders Psychology Synthesizers Distortion measurement Frequency measurement Speech synthesis Nonlinear filters Performance evaluation

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