Showing papers on "Code-excited linear prediction published in 1990"

Fast methods for the CELP speech coding algorithm

Abstract: Special fast procedures for the code excited linear predictive coding (CELP) algorithm have been developed to make implementation on modest hardware possible. The advantages, as well as the disadvantages, of the various fast procedures are discussed. A general formalism for the algorithm is developed, followed by the discussion of the individual procedures which are grouped according to their features. Along with the computational complexity of each procedure, its storage requirement and numerical accuracy are discussed. A large number of the fast procedures are designed to search through a particular type of codebook (most of the codebooks are stochastic in character, while a few are deterministic). Other fast procedures can be used for arbitrary codebooks and are thus also applicable to trained codebooks. Some of the fast procedures designed for stochastic codebooks can also be used for the computation of the closed pitch loop parameters, which can be interpreted as a search through a time-dependent codebook. >

On reducing computational complexity of codebook search in CELP coder through the use of algebraic codes

Abstract: A general framework is introduced which allows both fast search and freedom in designing codebooks with good statistical properties. Several previously proposed schemes are compared from this viewpoint. A backward filtering formulation is given to show that sparse algebraic codes (SACs) (i.e., with few nonzero components) offer distinct advantages. It is shown that they reduce the optimal-search computation per codeword. They also allow control of the statistical properties of the codebook in the time and frequency domains. This control can be dynamic in the sense that it can be made to evolve as a function of the linear predictive coding model A(z). The algebraic-code excited linear prediction (ACELP) technology which allows full duplex operation on a single TMS320C25 at rates between 4.8 and 16 kb/s and which is based on SAC-driven dynamic codebooks is described. >

Dynamic codebook for efficient speech coding based on algebraic codes

Abstract: A method of encoding a speech signal is presented. This method improves the excitation codebook and search procedure of the conventional Code Excited Linear Prediction (CELP) speech encoders. Use is made of a dynamic codebook (201, 202) based on the combination of two modules: a sparse algebraic code generator (201) associated to a filter (202) having a transfer function varying in time. The generator (102) is a structured codebook with codewords having very few non zero components. The filter (202) shapes the spectral characteristics whereby the resulting excitation codebook (201, 202) exhibits favorable perceptual properties. The search complexity in finding the best codeword is greatly reduced by bringing the search back to the algebraic code domain thereby allowing the sparsity of the algebraic code to speed up the necessary computations.

High-quality 16 kb/s speech coding with a one-way delay less than 2 ms

Abstract: A high-quality 16-kb/s speech coder which has a one-way coding delay of less than 2 ms is presented. The coder is basically a backward-adaptive version of the code-excited linear prediction (CELP) coder. The low coding delay is achieved by using backward-adaptive predictor and gain and by using an excitation vector size as small as five samples. The pitch predictor in conventional CELP coders is eliminated, and the linear predictive coding (LPC) predictor order is increased from 10 to 50. The excitation gain is updated by a tenth-order adaptive logarithmic gain predictor. This log-gain predictor and the LPC predictor are updated by performing LPC analysis on previous log-gain and coded speech, respectively. The excitation codebook is closed-loop optimized and the codebook index is Gray-coded to improve the robustness against channel errors. Formal subjective tests indicate that this coder produces high-quality speech comparable to that of the CCITT G.721 32-kb/s ADPCM standard. >

Improved pitch prediction with fractional delays in CELP coding

Abstract: A scheme is discussed for long-term prediction in CELP (code-excited linear predictive) coding using fractional delay prediction. This technique permits a more accurate representation of voiced speech and achieves an improvement of synthetic quality for female speakers. The higher complexity of this type of predictor relative to the classical one is its major disadvantage. Suboptimal schemes in which the search for the functional pitch delay is restricted to a neighborhood of an integer pitch estimate can be envisaged to decrease the computational load. >

4.8 kbit/s delayed decision CELP coder using tree coding

Abstract: A 4.8-kb/s delayed decision code excited linear prediction (CELP) coder that uses tree coding is described. In conventional CELP coding, short-term and long-term prediction parameters as well as excitation parameters are sequentially determined. In the proposed delayed decision CELP coding, a tree coding method is utilized. The long-term prediction and excitation parameter candidates obtained in each subframe are listed as a tree and the optimum combined parameter sequences are selected to minimize global quantization distortion over the coding frame. The proposed coding method significantly increases the quality of the 4.8-kb/s CELP coder at the cost of an additional 5-ms coding delay. >

Speech coding system and a method of encoding speech

Abstract: A speech coding system of the code excited linear prediction (CELP) type comprises means (24,26) for filtering digitised speech samples to form perceptually weighted speech samples. Entries in a one-dimensional codebook (110) comprising frame length sequences are filtered in a perceptually weighted synthesis filter (28) to form a one-dimensional filtered codebook (38). The filtered codebook entries are compared with the perceptually weighted speech signals to obtain a codebook index which gives the minimum perceptually weighted error when the speech is resynthesised. Using a one-dimensional codebook (110) reduces the amount of computation which is required compared to using a two-dimensional codebook.

Real-time implementation and performance of a 16 kb/s low-delay CELP speech coder

Abstract: A real-time implementation of the low-delay code-excited linear prediction (LD-CELP) speech coder based on the AT&T DSP32C is described. The performance of the coder is described. Recursive windowing is used to calculate the autocorrelation coefficients by dividing the Levinson-Durbin recursion into smaller parts and spreading the computation over several speech vectors. The LD-CELP encoder uses about 90% of the processor time of an 80-ns DSP32C, while the decoder uses about 40%. Hence, the coder is implemented on two DSP32Cs. For a single encoding with clear or noisy channels, the speech quality produced by this coder is either equivalent to or better than the CCITT G.721 standard 32-kb/s ADPCM. However, for multiple asynchronous encodings the ADPCM coders gives slightly better performance. The LD-CELP coder passes dual-tone multifrequency tones and 300-, 1200-, and 2400-b/s modem signals. >

CELP coding for land mobile radio applications

Abstract: The performance of the Proposed Federal Standard 1024 8000-b/s system for digital land mobile radio on a 12.5- or 6.25-kHz channel is described. The system contains the Proposed Federal Standard 1016 4800-b/s code excited linear predictive (CELP) voice coder developed by the US DOD and AT&T Bell Laboratories. The system is error protected with 2400 b/s and has 800 b/s for overhead signaling. The system is intended to replace the 12 kHz continuous variable slope deltamod (CVSD) voice coder which is used in Federal Standard 1025. The error protection technique used is Golay coding with hard or soft decoding. The performance of this error protection over simulated fading radio channels is described. >

Finite state CELP for variable rate speech coding

Abstract: A finite-state code excited linear prediction (CELP) system is proposed for variable-rate speech coding. The encoding system consists of a number of CELP coders with different linear predictive coding parameter quantization patterns, code book sizes, and population densities. The selection of the encoding state for each input vector depends on the input signal characteristics, the desired bit rate/signal to quantized noise ratio, and the current state of the encoder. In CELP coders the greater part of the bit resources, about 70%, is used for encoding of the excitation signal. However, the excitation accuracy needed to encode a speech segment with a desired level of fidelity strongly depends on its short-term spectral characteristics. The use of a finite-state system involves some implicit clustering of speech and allows variable-rate coding of the excitation. The gain in compression that can be obtained from variable-rate coding of the excitation signal is investigated. Experiments with a four-state variable-rate CELP coder produce good-quality encoded speech at 5 kbit/s. >

Pitch-orthogonal code-excited LPC

Abstract: An algorithm is proposed for the purpose of removing the coding redundancy between the adaptive codebook, or pitch, and the stochastic codebook, in a standard CELP (code-excited linear predictive) speech coder, allowing the calculation of an optimum two-vector gain set using sequential one-vector quantizers. This is done by orthogonalizing tile unweighted stochastic codebook to a codebook redundancy vector produced by backward-filtering the selected pitch vector. Using this algorithm, it is possible to design a reasonable-quality speech coder using the computationally-minimal hexagonal lattice algebraic codebook, or a high-quality embedded coder using multistage vector excitation. >

A low-delay code-excited linear predictive coder for speech or audio

Abstract: A code-excited linear-predictive (CELP) coder for speech or audio transmission at compressed (e.g., 16 kb/s) data rates is adapted for low-delay (e.g., less than 'five ms. per vector) coding by performing spectral analysis of at least a portion of a previous frame of simulated decoded speech to determine a synthesis filter of a much higher order than conventionally used for decoding synthesis and then transmitting only the index for the vector which produces the lowest internal error signal. In other contexts, performing less critical portions of analysis on a previous frame of data has been termed "backward-adaptive". Two types of gain adaptation are used in respective implementations, both of which are backward-adaptive and one of which is dynamic to the extent that weighting coefficients are iteration-dependent. The spectral analysis is twentieth-order to fiftieth-order linear prediction analysis. Pseudo-Gray coding is used to reduce errors induced by channel noise; and the effects of the gain and specrtral analyses are taken into account in the training procedures which determine the sets of vectors which are the reference or candidate excitation vectors.

Constrained-stochastic-excitation coding

Abstract: In CELP coding, stochastic (noise-like) excitation is used in exciting a cascade of long-term and short-term all-pole linear synthesis filters. This approach is based on the observation that the ideal excitation, obtained by inverse-filtering the speech signal, can be modeled for simplicity as Gaussian white noise. Although such stochastic excitation resembles the ideal excitation in its global statistical properties, it contains a noisy component that is irrelevant to the synthesis process. This component introduces some roughness and noisiness in the synthesized speech. The present invention reduces this effect by adaptively controlling the level of the stochastic excitation. The proposed control mechanism links the stochastic excitation to the long-term predictor in such a way that the excitation level is inversely related to the efficiency of the predictor. As a result, during voiced sounds, the excitation level is considerably attenuated and the synthesis is mainly accomplished by exciting the short-term filter with the periodic output of the long-term filter. This reduces the noisiness, enhances the pitch structure of the synthesized speech and its perceptual quality.

A system for coding wide-band audio signals

Abstract: A method of and apparatus for coding wide-band audio signals are provided, wherein the band of the signal to be coded is split into two sub-bands; the signals in each sub-band are coded in coders (CD1, CD2) operating according to analysis-by-­synthesis techniques and exploiting short-term and long-term correlations in the speech signal. Two embodiments of such coders are described, the first utilizing a multi-pulse excitation and the other the CELP coding technique.

High-quality digital speech at 4 kb/s

Abstract: A speech coder based on a single-pulse excitation code-excited linear predictive coding (SPE-CELP) model of linear-predictive coding (LPC) is proposed. An algorithm for determining the time instants of pitch periods within a short interval of periodic speech, which results in a time sequence of marker points that indicate the beginning of the pitch periods in the analyzed speech interval, is described. The LPC excitation is generated by a stochastic codebook for nonperiodic speech and by a single pulse per pitch period for periodic speech. The proper alignment of the excitation pulse is efficiently computed using dynamic programming. It is concluded that, at overall bit rates of around 3 kb/s, the coder produces significantly better speech quality than LPC10E, though the synthesized speech still sounds slightly buzzy for certain speakers. >

Fractal excitation signals for CELP speech coders

Abstract: A class of random signals is presented as a new excitation codebook for stochastic predictive speech coders. These signals, known as fractional noises, depend on a single parameter, 0

Low bit-rate spectral coding in CELP, a new LSP-method

Abstract: The use of line spectral pair (LSP)-frequencies for approaching the performance of a full vector quantizer for speech coding is discussed. An algorithm is presented that attains close to optimal performance if an accumulative constraint is introduced in the parameter encoding. Numerical results are presented from the training of code-books as well as from the evaluation of the quantizer. Some listening tests are discussed. >

Error detection and control for the parametric information in CELP coders

Abstract: Optimum quantization and code assignment schemes are described which minimize the subjective quality degradations introduced into the output speech of code excited linear predictive (CELP) coders by channel degradations. The background and basis for use of minimum redundancy for error control are examined. Greater emphasis is placed on adjustment of corrupted parameters to minimize subjective degradation rather than outright bit-by-bit error correction. These schemes are mostly tested on the CELP baseband coder, but they should be applicable to any linear predictive coder. They raise the bit rate of a 4.8-kb/s coder by about 12.5% and its MOS at 2*10/sup -2/ bit error rate by about 21.1% (scale 1-5). >

Low-rate analysis-by-synthesis wideband speech coding

Abstract: This thesis studies low-rate wideband analysis-by-synthesis speech coders. The wideband speech signals have a bandwidth of up to 8 kHz and are sampled at 16 kHz, while the target operating bit rate is 16 kbitslsec. Applications for such a coder range from high-quality voice-mail services to teleconferencing. In order to achieve a low operating rate, the coding places more emphasis on the lower frequencies (0 to 4 kHz), while the higher frequencies (4 to 8 kHz) are coded less precisely but with little perceived degradation. The study consists of three stages. First, aspects of wideband spectral envelope modeling using Line Spectral Frequencies (LSF's) are studied. Then, the underlying coder structure is derived from a basic Residual Excited Linear Predictive coder (RELP). This structure is enhanced by the addition of a pitch prediction stage, and by the development of full-band and split-band pitch parameter optimization procedures. These procedures are then applied to an Code Excited Linear Prediction (CELP) model. Finally, the performance of full-band and split-band CELP structures are compared.

CELP coding with an adaptive density pulse excitation model

Abstract: An approach to dynamic bit-allocation to excitation vectors to improve the performance of a code excited linear prediction (CELP) coder is proposed. The method is based on an adaptive density pulse (ADP) excitation. By using the ADP, bit-allocation to the excitation vector can be easily varied. Also, the number of samples of excitation can be reduced. The effects of the ADP parameters on the synthetic speech quality are discussed. The ADP-CELP coder is described. The benefit of introducing the ADP excitation model to the CELP coder is evaluated. The segmental signal-to-noise ratio gains of the ADP-CELP coder over the conventional CELP are about 2 dB both at 8 kbit/s and at 4.8 kbit/s. >

Structured codebook design in CELP

Abstract: Codebook Excited Linear Protection (CELP) is a popular analysis by synthesis technique for quantizing speech at bit rates from 4 to 6 kbps. Codebook design techniques to date have been largely based on either random (often Gaussian) codebooks, or on known binary or ternary codes which efficiently map the space of (assumed white) excitation codevectors. It has been shown that by introducing symmetries into the codebook, good complexity reduction can be realized with only marginal decrease in performance. Codebook design algorithms are considered for a wide range of structured codebooks.

Combined source and channel coding based on multimode coding

Abstract: A multimode source/channel coder which can dynamically control the balance of source and channel coding according to the channel quality is introduced. As an example of such a system, a 4.8-kb/s code excited linear predictive coder with three coding modes (A, B1, and B2), each of which has different bit assignments to source and channel coding, is presented. The optimum coding mode is selected in each frame, based on an evaluation of the spectral distortion (SN/sub LAR/) in reproduced speech. The threshold value of SN/sub LAR/ for mode decision is varied according to the channel error rate. Computer simulation shows that 2 dB and 3 dB of SNR/sub seg/ improvement is achieved at a bit error rate of 3*10/sup -3/ and 3*10/sup -2/ respectively, over conventional CELP without channel coding. >

A proposed Federal Standard for narrowband digital land mobile radio

Abstract: The technical characteristics and features of a standard defining interoperable digitized voice and digital data modes operating at 8000 b/s with 125 kHz or narrower channel spacings at frequencies above 30 MHz are described The voice modes will incorporate the US government standard 4800 b/s code-excited linear prediction (CELP) voice coding algorithm Both voice and data modes will utilize error control coding with interleaving to combat error bursts typical of digital land mobile radio communication channels The baseband transmission format is designed with a superframe structure that will provide for reliable late-entry synchronization The data rate, transmission modes, transmission format, and modulation technique are discussed >

Speech Coding Based On Vector Quantization For A Tactile Vocoder Hearing Device

Abstract: A portable digital speech processing system for deaf patients is described in the paper. This real-time processing system provides appropriate information to a ten-channel tactile prosthesis depending on speech signal caracteristics to be perceived. The system is based on three chips: a Codec TCM29C13, a DSP Digital Signal Processor TMS320E17 and a full custom integrated BiCmos circuit. Speech coding is based on LPC coefficients estimation and vector quantization. The architecture of this portable low power unit allows a marked improvement synthetic vowel discrimination.

Recent Trends and Techniques in Speech Coding

Abstract: Analysis-by-synthesis predictive coding of speech has recently become the dominant approach to efficient coding of speech at bit rates ranging from 4 to 16 kb/s. Some of the techniques already in use or currently being investigated include vector quantization, structured codebooks for reducing complexity, adaptive postfiltering, interframe predictive coding of speech parameters, backward adaptation for low delay coding, phonetic segmentation, adaptive postfiltering, and pseudo-Gray coding of channel symbols.

LD-CELP: a high quality 16 kb/s speech coder with low delay

Abstract: The coder is basically a backward-adaptive version of the code-excited linear prediction (CELP) coder, and is named low-delay CELP, or LD-CELP. The low coding delay (less than 2 ms) is achieved by using backward-adaptive predictor and gain, and by using a small excitation vector size. This coder has been implemented in real-time using the AT&T DSP32C floating-point DSP chips. Currently the encoder uses about 90% of the processor time while the decoder takes about 40%. Fixed-point implementation of the coder is also possible with some modifications of the algorithm. The speech quality of 16 kb/s LD-CELP is approximately equivalent to that of the CCITT G.721 32 kb/s ADPCM (adaptive differential PCM) standard. The coder can pass 300, 1200, and 2400 b/s modem signals as well as DTMF (dual-tone multifrequency) tones. From the preliminary results of CCITT laboratory tests, it appears that this coder can meet all CCITT requirements for the 16 kb/s speech coding standard. >

CELP coding with adaptive output-error model identification

Abstract: An approach to enhancing the code excited linear prediction (CELP) model for speech coding is presented. Adaptive infinite impulse response (IIR) filtering is used to reestimate synthesis filter parameters to minimize the output-error. Thus, not only the excitation sequence, but part of the filter is also computed in an analysis by synthesis framework. The method is extended to modeling voiced speech with a stylized glottal excitation. This approach to modeling voiced speech gives the potential of manipulating pitch while retaining a high quality. >

System for speech coding and apparatus for the same

Abstract: A CELP type of speech signal coding system, wherein a code vector obtained by applying linear prediction to a vector of a residual speech signal of white noise stored in a code book and a pitch prediction vector obtained by applying linear prediction to a residual signal of a preceding frame given a delay corresponding to a pitch frequency are added, use is made of an impulse vector obtained by applying linear prediction to a residual signal vector of impulses having a predetermined relationship with the vectors of the white noise code book, variable gains are given to at least the above code vector and impulse vector, a reproduced signal is produced, and this reproduced signal is used for identification of the input speech signal, thus enabling the creation of a pulse series corresponding to the sound source of voiced speech sounds, enabling accurate evaluation and identification of a pulse-like sound source of voiced speech sounds and enabling improvement of the quality of the reproduced speech while reducing the amount of information transmitted.