Showing papers on "Code-excited linear prediction published in 1987"

Fast CELP coding based on algebraic codes

Abstract: Code-Excited Linear Prediction (CELP) produces high quality synthetic speech at low bit rate. However the basic scheme leads to huge computational loads. The paper describes a related scheme, which allows real time implementation on current DSP chips. The very efficient search procedure in the codebook is achieved by means of a new technique called "backward filtering" and the use of algebraic codes. RSB performances are reported for a variety of conditions.

Quantization procedures for the excitation in CELP coders

Abstract: Past research on CELP (Code-Excited Linear Predictive) coders has mainly concentrated on the feasibility of the CELP concept and on the reduction of the computational complexity. In this paper we address the problem of finding and encoding the excitation parameters with a limited bit rate, such that high quality speech coding in the 4.8 - 7.2 kb/s range becomes feasible. First, we examine the effect of the various excitation parameters such as code book size, code book population, order of the long-term predictor and update rate on the quality of the reconstructed speech. Second, we investigate procedures for designing and incorporating quantizers for the parameters involved. Finally, using both scalar and vector quantization techniques for the LPC coefficients, we simulated 4.8 kb/s and 7.2 kb/s coders. We also report on the use of postfiltering to further improve the performance of the CELP coder.

Code excited linear predictive vocoder

Abstract: Apparatus for encoding speech using a code excited linear predictive (CELP) encoder using a recursive computational unit. In response to a target excitation vector that models a present frame of speech, the computational unit utilizes a finite impulse response linear predictive coding (LPC) filter and an overlapping codebook to determine a candidate excitation vector from the codebook that matches the target excitation vector after searching the entire codebook for the best match. For each candidate excitation vector accessed from the overlapping codebook, only one sample of the accessed vector and one sample of the previously accessed vector must have arithmetic operations performed on them to evaluate the new vector rather than all of the samples as is normal for CELP methods. For increased performance, a stochastically excited linear predictive (SELP) encoder is used in series with the adaptive CELP encoder. The SELP encoder is responsive to the difference between the target excitation vector and the best matched candidate excitation vector to search its own overlapping codebook in a recursive manner to determine a candidate excitation vector that provides the best match. Both of the best matched candidate vectors are used in speech synthesis.

A comparison of some algebraic structures for CELP coding of speech

Abstract: The paper discusses how binary error-correcting codes can be used to provide (+1/-1)-waveform codebooks that speed up search in CELP vocoders. Four coding techniques operating at half bit per sample with respectively 8, 16, 24 and 32 samples are compared in terms of complexity and SNR performance, Recent results on spherical codes from regular point lattices are also reported.

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

Abstract: 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.

Code excited linear predictive vocoder using virtual searching

Abstract: Apparatus for encoding speech using a code excited linear predictive (CELP) encoder using a virtual searching technique during speech transitions such as from unvoiced to voiced regions of speech. The encoder compares candidate excitation vectors stored in a codebook with a target excitation vector representing a frame of speech to determine the candidate vector that best matches the target vector by repeating a first portion of each candidate vector into a second portion of each candidate vector. For increased performance, a stochastically excited linear predictive (SELP) encoder is used in series with the adaptive CELP encoder. The SELP encoder is responsive to the difference between the target vector and the best matched candidate vector to search its own overlapping codebook in a recursive manner to determine a candidate vector that provides the best match. Both of the best matched candidate vectors are used in speech synthesis.

A new speech coding model based on a least-squares sinusoidal representation

Abstract: In recent years the concept of analysis-by-synthesis has been applied very successfully to improving the performance of LPC based models At the same time, new speech models have been introduced based on representing speech by a sum of amplitude and frequency-modulated sinusoids which have been shown to successfully represent the non-linear, time-varying and quasi-periodic nature of speech In this paper we present an approach to applying the analysis-by-synthesis technique to sinusoidal speech modelling in an attempt to increase the ability of the model to accurately represent the speech waveform

Quality comparison of low complexity 4800 bps self excited and code excited vocoders

Abstract: This paper presents a formal objective and subjective comparison of a number of LPC vocoders which operate at bit rates around 4800 bps. In this work, particular emphasis is placed on the Self Excited Vocoder (SEV), a new speech coding approach which was introduced by the authors at ICASSP86 [1]. Many members of a class of LPC vocoders of which the SEV, the well known Multiple Pulse Excited Linear Predictive Coder (MPLPC) [2], and Code Excited Linear Predictive Coder (CELPC) [3] are members, are simulated and compared. Through these experiments, the SEV is shown to be a low complexity, simply implemented speech coder that is competitive with the other coders in this class in producing high quality speech at low bit rates.

Covariance and autocorrelation methods for vector linear prediction

Abstract: A novel least-squares formulation of the vector linear prediction (VLP) problem is presented. Based on this formulation, we develop two new design methods for obtaining the optimal vector predictor for frame-adaptive prediction: the covariance method and the autocorrelation method, which bear the names of the corresponding methods in scalar LPC analysis. Our formulation reveals several previously unrecognized properties of the resulting normal equation. Simulation results for VLP of speech waveforms confirm that the two proposed methods indeed give higher prediction gain than previously developed methods.

A technique for using multipulse linear predictive speech synthesis in text-to-speech type systems

Abstract: A scheme for alteration of pitch and timing in multipulse linear predictive synthesized speech is presented. The method is a hybrid of waveform coding and linear prediction. It attempts to combine the high-quality sound units available in waveform techniques with the ability to alter prosody available from linear prediction.

Harmonic coding of speech at 4.8 kb/s

Abstract: This paper describes a new speech coding technique which yields improved speech quality over existing 2.4 kb/s LPC vocoders. The method is computationally efficient and operates at a data rate of 4.8 kb/s. Each speech frame is initially classified as voiced or unvoiced. Unvoiced frames are synthesized using a linear predictive coding filter with noise or multipulse excitation. Voiced frames are synthesized using a sum of sinusoids. The frequency of each sinusoid is defined by peaks in the frequency spectrum. A new interpolation technique provides a computationally efficient method of locating the spectral peaks. A real-time, fully quantized version has been implemented in hardware.

Apparatus for using a Leroux-Gueguen algorithm for coding a signal by linear prediction

Abstract: The invention relates to an apparatus for using the Leroux-Gueguen algorithm for coding a signal by linear prediction. Apparatus applicable to the coding of a signal uses the Leroux-Gueguen algorithm for coding the signal by linear prediction. The apparatus receives at one input, correlation, multiplexers, two multipliers, two adders and a divider which, by successive iterations, calculate intermediate variables making it possible to pass from correlation coefficients R i to PARCOR coefficients K i .

A new approach to noise-robust LPC

Abstract: In this paper a new spectral estimate of noisy speech derived from linear predictive Autoregressive(AR) analysis is compared with the spectra produced by single order Linear Predictive Coding(LPC). The new estimate consists of a bin-by-bin average cf a set of linear prediction AR spectra with orders between 10 and 30. According to simulation results the improvement in peak centre frequency and bandwidth estimation resulting from using this model for noisy speech is equivalent to increasing the SNR by about 5dB when the SNR of the input signal is 20dB or less.

Device for the execution of an algorithm (leroux-gueguen) for the coding of a signal by linear prediction

Abstract: The invention relates to an apparatus for using the Leroux-Gueguen algorithm for coding a signal by linear prediction. Apparatus applicable to the coding of a signal uses the Leroux-Gueguen algorithm for coding the signal by linear prediction. The apparatus receives at one input, correlation, multiplexers, two multipliers, two adders and a divider which, by successive iterations, calculate intermediate variables making it possible to pass from correlation coefficients Ri to PARCOR coefficients Ki.

Vector Predictive Coding for Very Low-Bit-Rate Speech Encoders

Abstract: The application of adaptive vector predictive coding for very low-bit-rate speech encoders was investigated and the performance quantified. The input speech samples were analyzed using the government standard LPC-10 algorithm to generate a vector with 10 LPC coefficients. This LPC vector was used as the input for vector prediction. Three prediction schemes: time-invariant prediction switched prediction and continuous prediction were investigated. The assumption of independence among the LPC coefficients was also studied. The results indicated the performance of time-invariant and independent prediction schemes is satisfactory for very-low-bit rate speech encoding applications.

Pole-zero analysis of speech by linear predictive coding.

Abstract: A method for generating pole-zero models for speech segments is described based on extracting Linear Predictive Coding (LPC) coefficients. Cepstral coefficient are generated from the LPCs and are used to obtain the negative derivative of the phase, thus enabling the spectrum to be split into a pole part and a zero part. Word classification by pole-zero tables, and their use for word recognition are described.

Information-theoretic compressibility of speech data

Abstract: Two standard reversible coding algorithms, Ziv-Lempel and a dynamic Huffman algorithm, are applied to various types of speech data. The data tested were PCM, DPCM, and prediction residuals from LPC. Neither algorithm shows much promise on small amounts of data, but both performed well on large amounts. Typically the Ziv-Lempel required about 12 seconds of data (with 8000 samples per second) to reach a stable compression rate. The dynamic Huffman coding took much less time to "warm up", often needing something like 64 milliseconds. Approximately 66 seconds of PCM with 12 bits per samples was compressed 6.4% by the Ziv-Lempel coding and 20.7% by the dynamic Huffman coding. The same numbers for DPCM with 13 bits per sample are 17.7% and 35.6% respectively. The prediction residuals had compression rates very close to those of DPCM, regardless of whether 1, 2, 5, or 10 prediction coefficients were used.