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Algebraic codebook search strategy for an algebraic code-excited linear-prediction speech coder by means of reduced candidate mechanism and iteration-free pulse replacement

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IET SIGNAL PROCESSING 2014年第9期8卷 990-995页

作者： Yeh, Cheng-Yu Natl Chin Yi Univ Technol Dept Elect Engn Taichung 41170 Taiwan

This work aims to present a combined version of reduced candidate mechanism (RCM) and iteration-free pulse replacement (IFPR) as a novel and efficient way to enhance the performance of algebraic codebook search in an algebraic code-excited linear-prediction speech coder. As the first step, individual pulse contribution in each track is given by RCM, and the value of N is then specified. Subsequently, the replacement of a pulse is performed through the search over the sorted top N pulses by IFPR, and those of 2-4 pulses are carried out by a standard IFPR. Implemented on a G.729A speech codec, this proposal requires as few as 20 searches, a search load tantamount to 6.25% of G.729A, 31.25% of the global pulse replacement method (iteration = 2), 41.67% of IFPR, but still provides a comparable speech quality in any case. The aim of significant search performance improvement is hence achieved in this work.

关键词： search problems vocoders speech coding iterative decoding algebraic codes linear predictive coding speech codecs algebraic codebook search strategy algebraic code excited linear prediction speech coder reduced candidate mechanism iteration free pulse replacement RCM IFPR standard IFPR G.729A speech codec global pulse replacement method search performance improvement

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Efficient coding of homogeneous textures using stochastic vector quantisation and linear prediction

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IEE PROCEEDINGS-VISION IMAGE AND SIGNAL PROCESSING 1999年第3期146卷 151-158页

作者： Gost, DG Torres, L Univ Politecn Cataluna Dept Signal Theory & Commun Barcelona Spain

Vector quantisation (VQ) has been extensively used as an effective image coding technique. One of the most important steps in the whole process is the design of the codebook. The codebook is generally designed using the LEG algorithm which uses a large training set of empirical data that is statistically representative of the images to be encoded. The LEG algorithm, although quite effective for practical applications, is computationally very expensive and the resulting codebook has to be recalculated each time the type of image to be encoded changes. Stochastic vector quantisation (SVQ) provides an alternative way for the generation of the codebook. In SVQ, a model for the image is computed first, and then the codewords are generated according to this model and not according to some specific training sequence. The SVQ approach presents good coding performance for moderate compression ratios and different type of images. On the other hand, in the context of synthetic and natural hybrid coding (SNHC), there is always need for techniques which may provide very high compression and high quality for homogeneous textures. A new stochastic vector quantisation approach using linear prediction which is able to provide very high compression ratios with graceful degradation for homogeneous textures is presented. Owing to the specific construction of the method, there is no block effect in the synthetised image. Results, implementation details, generation of the bit stream and comparisons with the verification model of MPEG-4 are presented which prove the validity of the approach. The technique has been proposed as a still image coding technique in the SNHC standardisation group of MPEG.

关键词： LBG algorithm still image coding technique VQ stochastic processes linear predictive coding linear prediction Computer vision and image processing techniques vector quantisation stochastic vector quantisation image coding homogeneous textures MPEG-4 codebook design image texture Image and video coding SVQ SNHC standardisation high compression ratios synthetic/natural hybrid coding bit stream generation

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DECOMPOSITION OF THE LPC EXCITATION USING THE ZINC BASIS FUNCTIONS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1989年第9期37卷 1329-1341页

作者： SUKKAR, RA LOCICERO, JL PICONE, JW TEXAS INSTRUMENTS INC DALLASTX 75265

A linear predictive coding (LPC) excitation signal composed of a set of orthogonal functions called zinc functions is introduced. These functions are shown to form a complete orthogonal set and have properties that are well suited for modeling the LPC residual signal. A benchmark comparison between Fourier series and zinc function modeling shows that the zinc function model for the residual is superior in the mean-squared-error sense. The zinc basis functions are used in two low-bit-rate speech coding systems targeted at the 4.8-9.6-kb/s range. The first is a zinc excited LPC (ZELPC) system, where the voiced excitation is modeled using the zinc functions while the unvoiced excitation is represented by the usual white noise source. The second system is a zinc multipulse LPC (ZMPLPC) system, where the LPC excitation is constructed using the zinc basis functions instead of the usual ideal impulses. Results show that, given a fixed segmental signal-to-noise ratio with similar computational complexity, the ZMPLPC system is more efficient than a conventional multipulse LPC (MPLPC) system. Subjective listening tests also indicate a preference for the ZMPLPC system.< >

关键词： linear predictive coding Zinc Fourier series Power harmonic filters Cutoff frequency Bit rate Speech coding White noise Vocoders Signal to noise ratio

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Optimal multistage vector quantization of LPC parameters over noisy channels

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2004年第1期12卷 1-8页

作者： Krishnan, V Anderson, DV Truong, KK Georgia Inst Technol Sch Elect & Comp Engn Atlanta GA 30332 USA Polycom Inc Pleasanton CA 94588 USA

The direct use of vector quantization (VQ) to encode LPC parameters in a communication system suffers from the following two limitations: 1) complexity of implementation for large vector dimensions and codebook sizes and 2) sensitivity to errors in the received indices due to noise in the communication channel. In the past, these issues have been simultaneously addressed by designing channel matched multistage vector quantizers (CM-MSVQ). A sub-optimal sequential design procedure has been used to train the codebooks of the CM-MSVQ. In this paper, a novel channel-optimized multistage vector quantization (CO-MSVQ) codec is presented, in which the stage codebooks are jointly designed. The proposed codec uses a source and channel-dependent distortion measure to encode line spectral frequencies derived from segments of a speech signal. Extensive simulation results are provided to demonstrate the consistent reduction in both the mean and the variance of the spectral distortion obtained using the proposed codec relative to the conventional sequentially designed CM-MSVQ. Furthermore, the perceptual quality of the reconstructed speech using the proposed codec was found to be better than that obtained using the sequentially designed CM-MSVQ.

关键词： channel coding linear predictive coding speech coding vector quantization

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A 6.7 kbps vector sum excited linear prediction on TMS320C54X digital signal processor

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MICROPROCESSORS AND MICROSYSTEMS 2002年第1期26卷 27-38页

作者： Kumar, KND Prabhu, KMM Sommen, PCW Indian Inst Technol Dept Elect Engn Madras 600036 Tamil Nadu India Eindhoven Univ Technol Dept Elect Engn NL-5600 MB Eindhoven Netherlands

In this paper, a 6.7-kbps vector sum excited linear prediction (VSELP) coder with less computational complexity is presented. A very efficient VSELP codebook with nine basis vectors and a heuristic K-selection method (to reduce the search space and complexity) is constructed to obtain the stochastic codebook vector. The nine basis vectors are obtained by optimizing a set of randomly generated basis vectors. During the optimization process, we have trained the basis vectors to give the system apriori knowledge of the characteristics of the input. The coder is implemented on a TMS320C541 digital signal processor. The performance is evaluated by testing the 6.7-kbps VSELP coder with different test speech data taken from different speakers. The quality of the coder is estimated by comparing the performance of the 6.7-kbps VSELP coder with an 8-kbps VSELP speech coder based on the IS-54 standards. (C) 2002 Elsevier Science B.V. All rights reserved.

关键词： vector sum excited linear prediction code excited linear prediction linear predictive coding digital signal processor

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SHORT-TERM TEMPORAL DECOMPOSITION AND ITS PROPERTIES FOR SPEECH COMPRESSION

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第6期39卷 1282-1290页

作者： CHENG, YM OSHAUGHNESSY, D MCGILL UNIV DEPT ELECT ENGNMONTREAL H3A 2T5QUEBECCANADA

This paper deals with a technique of temporal speech compression, called temporal decomposition. A mathematical presentation of temporal decomposition is discussed from the point of view of speech production. We theoretically study temporal decomposition in terms of an iterative solution and its convergence. For vocoder applications, we propose a practical algorithm, short-term temporal decomposition. The statistical performance of this algorithm was tested on a multispeaker database. We comparatively study the efficiencies of vector quantization and vector predictability on data derived from the algorithm and on data from LPC analysis. The results show that data from the algorithm have more efficient information transmission via vector quantization than data from LPC, and that vector prediction coding is feasible for the data from temporal decomposition.

关键词： Speech coding Vocoders Iterative algorithms Vector quantization linear predictive coding Speech analysis Interpolation Matrix decomposition Distortion measurement Testing

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A FEEDBACK-CONTROL SYSTEM FOR REAL-TIME FORMANT ESTIMATION .2. ANALYSIS OF A HYSTERESIS EFFECT AND F2 ESTIMATION

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IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1993年第9期40卷 892-898页

作者： ZIERHOFER, CM HOCHMAIR, ES Institut für Experimentalphysik Universität Innsbruck Innsbruck Austria

This paper presents the second part of the analysis of a feedback control system for real-time formant estimation. The system behavior is analyzed for an input signal composed of two sinusoids. If the frequency difference between the two input spectral lines is sufficiently great and the amplitude ratio is within certain limits, a hysteresis effect occurs. Then the system shows a tendency to select one of the two input spectral lines. The existence of the second line has only little influence on the accuracy of the detection of the selected line. From the analysis, conclusions of the system behavior regarding formant estimation can be drawn. A design example for second formant detection is simulated and compared with the results obtained by simulation of a zero-crossing system for F2 estimation and LPC analysis.

关键词： Feedback control Real time systems Hysteresis Analytical models predictive models linear predictive coding Signal analysis Frequency

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Design and implementation of a parametric speech coder

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1998年第1期44卷 163-169页

作者： Kwong, S Nui, PT City Univ Hong Kong Dept Comp Sci Hong Kong Hong Kong

Currently, it has been found out that speech coders based on linear predictive coding are good and accurate in modeling speech utterances. It does not only modeled speech utterance in an accurate manner but it also has the following properties: i) it provides a good estimate of the vocal tract spectral envelope;ii) it is analytically tractable;iii) it can be implemented in either software or hardware;iv) it uses lesser data storage than many other approaches of speech coders. In this paper, a hybrid speech coder based on Code Excited linear predictive coding (CELPC) and Voice Excited linear predictive coding (VELPC) is presented. CELPC is currently one of the main techniques for producing high quality speech at around 4.8 kbps. However, the computation requirement for the original CELPC was too demanding and not suitable for practical use. Therefore, a hybrid approach to produce speech signals with high quality and with a bit rate of 3.478 kbps has been proposed. This method splits the speech signal into two portions, the base-band signal and the high-band signal in the frequency domain. The base-band and the high-band signal are then coded using the CELPC and VELPC techniques, respectively.

关键词： Speech coding linear predictive coding Signal processing Speech analysis Nonlinear filters predictive models Bit rate Quantization Encoding Computer science

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Joint quantisation strategies for low bit-rate sinusoidal coding

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IET SIGNAL PROCESSING 2010年第5期4卷 548-559页

作者： Unver, E. Villette, S. Kondoz, A. Univ Surrey Ctr Commun Syst Res Guildford GU2 7XH Surrey England

Although there are speech coding standards producing high-quality speech above 4 kbps, below that transparent quality has not been achieved yet. There is still room for improvement at lower bit rates, especially at 2.4 kbps and below, which is an area of interest for military and security applications. Strategies for achieving high-quality speech using sinusoidal coding at very low bit rates are discussed. Previous work in the literature on combining several frames in a metaframe and performing variable bit allocation within the metaframe is extended. Experiments have been carried out to find an optimum metaframe size compromise between delay and quantisation gains. Metaframe classification and quantisation according to the metaframe class are used for better efficiency. A method for voicing determination from the linear prediction coefficient (LPC) shape is also presented. The proposed techniques have been applied to the SB-LPC vocoder to produce speech at 1.2 and 0.8 kbps, and compared to the original SB-LPC vocoder at 2.4/1.2 kbps as well as an established standard (Mixed Excitation linear predictive - MELP - vocoder) at 2.4/1.2/0.6 kbps in a listening test. It has been found that the proposed techniques have been effective in reducing the bit rate while not compromising the speech quality.

关键词： low bit-rate sinusoidal coding metaframe classification speech coding Speech processing techniques speech quality linear prediction coefficient linear predictive coding Speech and audio coding mixed excitation linear predictive vocoder SB-LPC vocoder metaframe quantisation strategies Codecs, coders and decoders vocoders

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GLOTTAL INVERSE FILTERING BY JOINT ESTIMATION OF AN AR SYSTEM WITH A linear INPUT MODEL

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1986年第1期34卷 28-42页

作者： MILENKOVIC, P WAISMAN CTR MENTAL RETARDAT & HUMAN DEV SPEECH MOTOR CONTROL LABSMADISONWI

Glottal inverse filtering is a technique by which the flow past the time varying glottal constriction is estimated by a filtering operation on the acoustic signal in human speech. The filtering operation removes the effects of the vocal tract resonances to reveal the underlying voice source signal. A linear model of a glottal pulse waveform is described along with a procedure for jointly determining an AR model of the vocal tract response together with the parameters of the glottal pulse model. This technique is applied to inverse filtering in human speech and the results are compared to covariance method LPC analysis.

关键词： Nonlinear filters Filtering Speech analysis Pulse shaping methods Speech synthesis Shape Resonance linear predictive coding Algorithm design and analysis Humans

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