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A new phase model for sinusoidal transform coding of speech

IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1998年第5期6卷 495-501页

作者： Ahmadi, S Spanias, AS Nokia Mobile Phones Inc San Diego CA 92121 USA

A phase modeling algorithm for sinusoidal analysis-synthesis of speech is presented, where short-time sinusoidal phases are approximated using a combination of linear prediction, spectral sampling, delay compensation, and phase correction techniques. The algorithm is different than phase compensation methods proposed for source-system LPC in that it has been tailored to sinusoidal representation of speech. Performance analysis on a large speech data base reveals improvement in temporal and spectral signal matching, as well as in the subjective quality of reconstructed speech. The method can be applied to enhance phase matching in low bit rate sinusoidal coders, where underlying sine wave amplitudes are extracted from an all-pole model. Preliminary subjective results are presented for a 2.4 kb/s sinusoidal coder.

关键词： linear prediction linear predictive coding phase modeling sinusoidal representation speech coding

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LPC SPEECH coding BASED ON VARIABLE-LENGTH SEGMENT QUANTIZATION

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1988年第9期36卷 1437-1444页

作者： SHIRAKI, Y HONDA, M Speech Research Group of the Basic Research Laboratories NTT Musashino Tokyo Japan

A low-bit-rate linear predictive coder (LPC) that is based on variable-length segment quantization is presented. In this vocoder, the speech spectral-parameter sequence is represented as the concatenation of variable-length spectral segments generated by linearly time-warping fixed-length code segments. Both the sequence of code segments and the segment lengths are efficiently determined using a dynamic programming procedure. This procedure minimizes the spectral distance measured between the original and the coded spectral sequence in a given interval. An iterative algorithm is developed for designing fixed-length code segments for the training spectral sequence. It updates the segment boundaries of the training spectral sequence using an a priori codebook and updates the codebook using these segment sequences. The convergence of this algorithm is discussed theoretically and experimentally. In experiments, the performance of variable-length segment quantization for voice coding is compared to that of fixed-length segment quantization and vector quantization.< >

关键词： linear predictive coding Speech coding Quantization Algorithm design and analysis Iterative algorithms Vocoders Speech processing Signal processing algorithms Bit rate Convergence

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Design and Performance of an Algorithm for Estimating Vocal System Parameters

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第4期2卷 531-536页

作者： Thomson, Mark M. Guillemin, Bernard J. Univ Auckland Dept Elect & Elect Engn Auckland 1 New Zealand

In this correspondence, we present two findings regarding an algorithm proposed by Milenkovic for determining the source and transfer function of the human vocal system. First, we show that more reliable estimates of the glottal endpoints can be obtained by modifying the procedure used to update the initial endpoint estimates. Second, we show that the algorithm is useful only for analyzing sounds of low to moderate pitch, because of its dependence on an initial transfer function estimate obtained by linear prediction.

关键词： Algorithm design and analysis Parameter estimation Transfer functions Least squares approximation Speech linear predictive coding Shape control Taylor series Polynomials Pulse shaping methods

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Strategies to improve the performance of very low bit rate speech coders and application to a variable rate 1.2 kb/s codec

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IEE PROCEEDINGS-VISION IMAGE AND SIGNAL PROCESSING 2005年第1期152卷 74-86页

作者： de Lamare, RC Alcaim, A CETUC PUC RIO BR-22453900 Rio De Janeiro Brazil

This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfifter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder.

关键词： speech synthesis speech codecs Speech and audio coding switched-predictive vector quantiser technique speech coding vocoders synthesised speech Codecs, coders and decoders ITU-T P.862 standard linear predictive coding decoded speech improvement noise reduction postfilter coding noise reduction 1.2 Kbit/s Filtering methods in signal processing vector quantisation 0.4 Kbit/s postfiltering approach LSF parameters mean opinion score mixed multiband excitation linear predictive coding MMBE approach variable rate speech codec Speech processing techniques LPC very low bit rate speech coder filtering theory perceptual evaluation of speech quality unvoiced frames low bit rate codec spectral envelope restoration

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Nonlinear predictive image coding with a neural network

Nonlinear predictive image coding with a neural network

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

作者： Z. He H. Li DSP Division Radio Engineering Department South-East University Nanjing China

A multilayer neural network used for nonlinear predictive image coding is described. Two coding schemes, nonadaptive and adaptive, are shown. Owing to matching of the local properties of the image, nonlinear predictive coding gives a better performance than linear predictive coding. A series of computer experiments shows the method has not only the ability to generalize but also noise reduction capabilities. Compared with differential pulse code modulation (DPCM), it greatly reduces the number of bits to be transmitted.< >

关键词： Image coding Neural networks Multi-layer neural network Pulse modulation predictive coding linear predictive coding Noise reduction Modulation coding

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The Modular Neural predictive coding architecture

The Modular Neural Predictive Coding architecture

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International Conference on Neural Information Processing

作者： M. Chetouani B. Gas J.L. Zarader Laboratoire des Instruments et Systemes d'Ile de France (LISIF) Universite Paris VI Paris France

ISBN: (纸本)9810475241

We present a new architecture called the Modular Neural predictive coding architecture (Modular NPC). This architecture is used for speech discriminant feature extraction (DFE). We present an application of the modular NPC architecture on phoneme recognition task. The phonemes which are extracted from the Darpa-Timit speech database are: vowels, /b/-/d/-/g/ and /p/-/t/-/k/ phonemes. Comparisons with coding methods (LPC, MFCC, PLP) are presented.

关键词： predictive coding Feature extraction linear predictive coding Speech recognition predictive models Mel frequency cepstral coefficient linearity Optimized production technology Speech processing Vectors

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Vector predictive quantization of the spectral parameters for low rate speech coding

Vector predictive quantization of the spectral parameters fo...

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

作者： Y. Shoham Dpt. of Signal Processing Research AT and T Bell Laboratories Inc. Murray Hill NJ USA

Vector predictive Quantization (VPQ) is proposed for coding the short-term spectral envelope of speech. The proposed VPQ scheme predicts the current spectral envelope from several past spectra, using a predictor codebook. The residual spectrum is coded by a residual codebook. The system operates in the log-spectral domain using a sampled version of the spectral envelope. Experimental results indicate a prediction gain in the range of 9 to 13 dB and an average log-spectral distance of 1.3 to 1.7 dB. Informal listening tests suggest that replacing the conventional scalar quantizer in a 4.8 Kbits/s CELP coder by a VPQ system allows a reduction of the rate assigned to the LPC data from 1.8 Kbits/s to 1.0 Kbits/s without any obvious difference in the perceptual quality.

关键词： Quantization Speech coding Vectors Speech analysis linear predictive coding Delay Switches Signal processing System testing Nonlinear filters

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DETECTION OF LARYNGEAL FUNCTION USING SPEECH AND ELECTROGLOTTOGRAPHIC DATA

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IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1992年第1期39卷 19-25页

作者： CHILDERS, DG BAE, KS Department of Electrical Engineering Mind-Machine Interaction Research Center University of Florida Medical School Gainesville FL USA Department of Electrical Engineering Kyungpook National University Taejon South Korea

The purpose of this research was to develop quantitative measures for the assessment of laryngeal function using speech and electroglottographic (EGG) data. We developed two procedures for the detection of laryngeal pathology: 1) a spectral distortion measure using pitch synchronous and asynchronous methods with linear predictive coding (LPC) vectors and vector quantization (VQ) and 2) analysis of the EGG signal using time interval and amplitude difference measures. The VQ procedure was conjectured to offer the possibility of circumventing the need to estimate the glottal volume velocity waveform by inverse filtering techniques. The EGG procedure was to evaluate data that was "nearly" a direct measure of vocal fold vibratory motion and thus was conjectured to offer the potential for providing an excellent assessment of laryngeal function. A threshold based procedure gave 75.9 and 69.0% probability of pathological detection using procedures 1) and 2), respectively, for 29 patients with pathological voices and 52 normal subjects. The false alarm probability was 9.6% for the normal subjects.

关键词： Speech Distortion measurement Pathology Time measurement linear predictive coding Vector quantization Signal analysis Filtering Motion measurement Vibration measurement

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REDUCTION OF BROAD-BAND NOISE IN SPEECH BY TRUNCATED QSVD

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1995年第6期3卷 439-448页

作者： JENSEN, SH HANSEN, PC HANSEN, SD SORENSEN, JA TECH UNIV DENMARK INST ELECTRDK-2800 LYNGBYDENMARK

We consider an algorithm for reduction of broadband noise in speech based on signal subspaces. The algorithm is formulated by means of the quotient singular value decomposition (QSVD). With this formulation, a prewhitening operation becomes an integral part of the algorithm. We demonstrate that this is essential in connection with updating issues in real-time recursive applications. We also illustrate by examples that we are able to achieve a satisfactory quality of the reconstructed signal.

关键词： Noise reduction Speech enhancement linear predictive coding Acoustic noise Telephony Microphones Noise cancellation Speech analysis Speech synthesis Matrix decomposition

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SHAPE-GAIN MATRIX QUANTIZERS FOR LPC SPEECH

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1986年第6期34卷 1427-1439页

作者： TSAO, C GRAY, RM STANFORD UNIV DEPT ELECT ENGNINFORMAT SYST LABSTANFORDCA 94305

It has been recently demonstrated that the principles of vector quantization for LPC speech can be simply extended to encompass matrices of LPC vectors with significant savings in bit rate. Unfortunately, however, such locally optimal matrix quantizers have prohibitively high complexity and memory requirements when implemented in a speech vocoder at bit rates giving acceptable quality speech. One approach to solving the problem is to separately code gain and shape in the matrix quantizer. This paper generalizes the principles of shape-gain vector quantizer design for LPC speech to matrix quantization and investigates the properties of the resulting quantizers. In particular, we present a design which combines shape matrices consisting of N shape vectors with K-dimensional gain vectors, where N and K are small integers, in practice, with K \geq N . Experimental results show that with K, N \geq 3 , significant reductions in bit rate over locally optimal vector quantizers are obtained for comparable performance. Simulations indicate that a shape-gain matrix quantizer, using a 10 bit shape codebook and an 8 bit codebook with K = N = 3 operating at 6 bits/frame for the LPC model, gives speech quality comparable to a locally optimal vector quantizer at 9 bits/frame. The matrix quantizer has somewhat greater than 5.7 times the memory requirement of the above vector quantizer, but less than 2.1 times the complexity. Subjective tests show that the speech from this matrix quantizer is intelligible to native speakers of English.

关键词： linear predictive coding Speech Shape Bit rate Information systems Signal processing algorithms Testing Government Laboratories

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