Algebraic Code Excited Linear Prediction Research Articles

A Comparative Study of Speech Coding Techniques for Electro Larynx Speech Production

Speech coding is a method of earning a tight speech signals representation for efficient storage and efficient transmission over band limited wired or wireless channels. This is usually achieved with acceptable representation and least number of bits without depletion in the perceptual quality. A number of speech coding methods already developed and various speech coding algorithms for speech analysis and synthesis are used. This paper deals with the comparison of selected coding methods for speech signal produced by Electro Larynx (EL) device. The latter is a device used by cancer patients with their vocal laryngeal cords being removed. The used methods are Residual-Excited Linear Prediction (RELP), Code Excited Linear Prediction (CELP), Algebraic Code Excited Linear Predictive (ACELP), Phase Vocoders based on Wavelet Transform (PVWT), Channel Vocoders based on Wavelet Transform (CVWT), and Phase vocoder based on Dual-Tree Rational-Dilation Complex Wavelet Transform (PVDT-RADWT). The aim here is to select the best coding approach based on the quality of the reproduced speech. The signal used in the test is speech signal recorded either directly by normal persons or else produced by EL device. The performance of each method is evaluated using both objective and subjective listening tests. The results indicate that PVWT and ACELP coders perform better than other methods having about 40 dB SNR and 3 PESQ score for EL speech and 75 dB with 3.5 PESQ score for normal speech, respectively.

A Secure AMR Fixed Codebook Steganographic Scheme Based on Pulse Distribution Model

Adaptive multi-rate (AMR), a popular audio compression standard, is widely used in mobile communication and mobile Internet applications and has become a novel carrier for hiding information. To improve the statistical security, this paper presents a steganographic scheme in the AMR fixed codebook (FCB) domain based on the pulse distribution model (PDM-AFS), which is obtained from the distribution characteristics of the FCB value in the cover audio. The pulse positions in stego audio are controlled by message encoding and random masking to make the statistical distribution of the FCB parameters close to that of the cover audio. The experimental results show that the statistical security of the proposed scheme is better than that of the existing schemes. Furthermore, the hiding capacity is maintained compared with the existing schemes. The average hiding capacity can reach 2.06 kbps at an audio compression rate of 12.2 kbps, and the auditory concealment is good. To the best of our knowledge, this is the first secure AMR FCB steganographic scheme that improves the statistical security based on the distribution model of the cover audio. This scheme can be extended to other audio compression codecs under the principle of algebraic code excited linear prediction (ACELP), such as G.723.1 and G.729.

A Search Complexity Improvement of Vector Quantization to Immittance Spectral Frequency Coefficients in AMR-WB Speech Codec

An adaptive multi-rate wideband (AMR-WB) code is a speech codec developed on the basis of an algebraic code-excited linear-prediction (ACELP) coding technique, and has a double advantage of low bit rates and high speech quality. This coding technique is widely used in modern mobile communication systems for a high speech quality in handheld devices. However, a major disadvantage is that a vector quantization (VQ) of immittance spectral frequency (ISF) coefficients occupies a significant computational load in the AMR-WB encoder. Hence, this paper presents a triangular inequality elimination (TIE) algorithm combined with a dynamic mechanism and an intersection mechanism, abbreviated as the DI-TIE algorithm, to remarkably improve the complexity of ISF coefficient quantization in the AMR-WB speech codec. Both mechanisms are designed in a way that recursively enhances the performance of the TIE algorithm. At the end of this work, this proposal is experimentally validated as a superior search algorithm relative to a conventional TIE, a multiple TIE (MTIE), and an equal-average equal-variance equal-norm nearest neighbor search (EEENNS) approach. With a full search algorithm as a benchmark for search load comparison, this work provides a search load reduction above 77%, a figure far beyond 36% in the TIE, 49% in the MTIE, and 68% in the EEENNS approach.

Efficient statistics-based algebraic codebook search algorithms derived from RCM for an ACELP speech coder

Applied to analgebraic codebook search conducted on an algebraic code-excitedlinear-prediction (ACELP) speech coder, two improved versions of reducedcandidate mechanism (RCM), designated as Fixed-G1-RCM and Fixed-2Track-RCM, arepresented in this study for further search performance improvement. It ismainly derived from two major research findings in a piece of our prior work.The first finding is that a pulse with a high contribution in a track is morelikely to serve as the optimal pulse in the optimal vector pertaining to thetrack, and the second is that the speech quality can be well maintained at asearch accuracy above 50%. In this proposal, the trade off can be tuned betweenthe search accuracy and the search complexity so as to reach a nearlyconsistent speech quality. It is experimentally demonstrated that either Fixed-G1-RCM(N = 3), or Fixed-2Track-RCM (M = 2, N = 3, or M = 3, N = 4) can provide ahighly superior search performance relative to a global pulse replacementmethod (iteration = 2) and an iteration-freepulse replacement method.DOI: http://dx.doi.org/10.5755/j01.itc.44.4.9847

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

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.

An efficient search algorithm for ACELP algebraic codebook

In a bid to enhance the search performance, this paper presents an improved version of reduced candidate mechanism (RCM), an algebraic codebook search conducted on an algebraic code‐excited linear‐prediction (ACELP) speech coder. This improvement is made on the basis of two findings in our prior work. The first finding is that a pulse with a high contribution in the associated track is more likely to serve as the optimal pulse in the optimal codevector; the second is that the speech quality can be well maintained at a search accuracy above 50% approximately. Subsequently, a new finding in this study is that there is a 0.8321 probability that the No. 1 ranked pulse in a global sorting by pulse contribution is indeed one of the optimal pulses. Hence, the No. 1 pulse in the global sorting is labeled as one of the optimal pulses, following which a sequence of search tasks is fulfilled through RCM. This proposed complexity reduction algorithm, implemented on a G.729A speech codec, takes as few as eight searches, a search load amounting to 2.5% of G.729A, 12.5% of global pulse replacement method (iteration = 2), 16.7% of iteration‐free pulse replacement method, and 50% of RCM (N = 2). This proposal is thus found to successfully reduce the required computational complexity to a great extent as intended. © 2014 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.

An efficient algebraic codebook search for ACELP speech coder

Abstract In a bid to enhance the search performance, this paper presents an improved version of reduced candidate mechanism (RCM), an algebraic codebook search conducted on an algebraic code-excited linear prediction (ACELP) speech coder. This improvement is made based on two findings in a piece of our prior work. The first finding is that a pulse with a high contribution in the associated track is more likely to serve as an optimal pulse in the optimal codevector and the second is that the speech quality can be well maintained at a search accuracy above 50% approximately. Subsequently, a new finding in this study concerning a structured algebraic codebook in G.729 indicates that there is a 0.8321 probability that the number 1 ranked pulse in a global sorting by pulse contribution is indeed one of the optimal pulses. Hence, the number 1 pulse in the global sorting is labeled as one of the optimal pulses, following which a sequence of search tasks are fulfilled through RCM. This proposed complexity reduction algorithm, implemented on a G.729A speech codec, takes as few as eight searches, a search load tantamount to 2.5% of G.729A, 12.5% of global pulse replacement method (iteration = 2), 16.7% of iteration-free pulse replacement method, and 50% of RCM (N = 2). This proposal is thus found to successfully reduce the required computational complexity to a great extent as intended.

An Efficient Search Strategy for ACELP Algebraic Codebook by Means of Reduced Candidate Mechanism and Iteration-Free Pulse Replacement

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 (ACELP) speech coder. As the first step, individual pulse contribution in each track is given by RCM, and the number N of candidate pulses 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 to 4 pulses are carried out by a standard IFPR. Implemented on a G.729A speech codec, this proposal requires as few as 24 searches, a search load tantamount to 7.5% of G.729A, 37.5% of the global pulse replacement method (iteration=2), 50% 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. DOI: http://dx.doi.org/10.5755/j01.itc.43.2.4946

Modification of codebook search in adaptive multi-rate wideband speech codecs using intelligent optimization algorithms

In recognition of high-quality wideband speech codecs, several standardization activities have been conducted, resulting in the selection of a wideband speech codec called adaptive multi-rate wideband (AMR-WB). The algebraic code-excited linear prediction (ACELP) technique is recommended in AMR-WB, and it is noted that most of the complexity in the ACELP structure comes from the codebook search. In this paper, a new method is proposed for codebook search based on the behavior of backward filtered target signal, d(n), introduced in ITU-T G.722.2 recommendation. To optimize the proposed scheme, five optimization algorithms (i.e., modified genetic algorithm, particle swarm optimization with dynamic inertia weight, bee colony optimization, modified differential evolution, and imperialist competition algorithm) are investigated. Experimental results show that the reduction in codebook search operations of the proposed method is able to reach up to 59 percent as compared with ITU-T G.722.2 recommendation. Meanwhile, BCO-based codebook search scheme has better convergence speed without significant degradation in quality metrics, such as segmental signal-to-noise ratio, mean opinion score, and perceptual evaluation of speech quality, when used in an AMR-WB speech codec.

Unified pulse-replacement search algorithms for algebra codebooks of speech coders

The algebraic code excited linear prediction (ACELP) algorithm, because of low complexity and high quality in its analysis-by-synthesis optimisation, has been adopted by many speech coding standards. This study proposes the unified generalised pulse replacement (UPR) search algorithm for the stochastic codebook of ACELP speech coders. The proposed UPR algorithm discusses the search breadth, the order of the search direction and the update frequency based on the pulse replacement method. In addition, there are many derivative types of UPR algorithms discussed. The proposed approaches can achieve the lowest computational complexity with imperceptible degradation of the speech quality. Furthermore, the normalised degradation ratio based on the standard subjective quality measurement is proposed to fairly compare the performance. The experimental results will verify the claims.

Constrained Voicing-Based Codebook in Low-Rate Wideband CELP Coding

In this work we propose an efficient technique to quantize the immitance spectral frequency (ISF) in an algebraic code-excited linear prediction (ACELP) wideband codec. The Constrained Voicing-Based Vector Quantization (C-VBVQ) presented in this paper improves substantially the performance of the unconstrained-VBVQ approach for both clean and noisy speech. Both techniques reduce the codebook search time by almost one third. However, in the C-VBVQ training phase, the three codebooks that are individually designed for voiced, unvoiced and transition speech, are jointly reoptimized to impose a structural configuration. The proposed technique reduces the processing delay since it restricts the quantization of an input vector to only one smaller but optimal codebook. For each speech frame, one codebook is selected from the set of three codebooks based on the interframe correlation of the spectral information. The C-VBVQ was successfully implemented in an ACELP wideband coder. The objective and subjective performance are not only superior to that of the combination of the split vector quantization and multistage vector quantization but also to the unconstrained VBVQ.

A semi-open-loop coding mode selection algorithm based on EFM and selected AMR-WB+ features

To solve the problems of the AMR-WB+ (Extended Adaptive Multi-Rate-WideBand) semi-open-loop coding mode selection algorithm, features for ACELP (Algebraic Code Excited Linear Prediction) and TCX (Transform Coded eXcitation) classification are investigated. 11 classifying features in the AMR-WB+ codec are selected and 2 novel classifying features, i.e., EFM (Energy Flatness Measurement) and stdEFM (standard deviation of EFM), are proposed. Consequently, a novel semi-open-loop mode selection algorithm based on EFM and selected AMR-WB+ features is proposed. The results of classifying test and listening test show that the performance of the novel algorithm is much better than that of the AMR-WB+ semi-open-loop coding mode selection algorithm.

The Implementation and Optimization of AMR on Mobile Device

C onsidering the limited resource of mobile device , the operation complexit y of the applications need s to be reduced. This paper presents the implementation of Adaptive Multi Rate (AMR) on mobile device and especially focuses on reducing the computational complexity. AMR is an algebraic code excited linear prediction (ACELP) based algorithm which could achieve satisfying speech quality by using the analysis-by-synthesis method. It has been chosen by the Third Generation Partnership Project (3GPP) as the mandatory codec for the third generation (3G) cellular systems. The optimizations include algorithm and instruction level. Based on the mobile device i-mate JAMin which has the ARM processor operating at 195 MHz, one second speech costs only 0. 6 second processing time. So, more resource is reserved for other applications.

Switching Search Method for Pulse Assignment in ITU-T G.729D

In this paper, the simplified search designs for the stochastic codebook of algebraic code excited linear prediction (ACELP) for ITU-T G.729D speech coder are proposed. By using two search rounds and limiting the search range, the computational complexity of the proposed approach is only 6.25% of the full search method recommended by G.729D. In addition, the computational complexity of proposed approach is only 59% of the global pulse replacement search method recommended by G.729.1. Simulation results show that the coded speech quality evaluated by using the standard subjective and objective quality measurements is with perceptually negligible degradation.

Compression of surface EMG signals with algebraic code excited linear prediction

Despite the interest in long timescale recordings of surface electromyographic (EMG) signals, only a few studies have focused on EMG compression. In this paper we investigate a lossy coding technique for surface EMG signals that is based on the algebraic code excited linear prediction (ACELP) paradigm, widely used for speech signal coding. The algorithm was adapted to the EMG characteristics and tested on both simulated and experimental signals. The coding parameters selected led to a compression ratio of 87.3%. For simulated signals, the mean square error in signal reconstruction and the percentage error in average rectified value after compression were 11.2% and 4.90%, respectively. For experimental signals, they were 6.74% and 3.11%. The mean power spectral frequency and third-order power spectral moment were estimated with relative errors smaller than 1.23% and 8.50% for simulated signals, and 3.74% and 5.95% for experimental signals. It was concluded that the proposed coding scheme could be effectively used for high rate and low distortion compression of surface EMG signals. Moreover, the method is characterized by moderate complexity (approximately 20 million instructions/s) and an algorithmic delay smaller than 160 samples (∼160 ms).

Efficient algebraic code-excited linear predictive codebook search

To reduce the computational complexity of algebraic code-excited linear prediction (ACELP) coders, an efficient codebook search mechanism based on a simplified correlation matrix (SCM) of the vocal impulse response is proposed. In the proposed approach, the statistical characteristics of the vocal impulse response are identified such that only a small proportion of the total number of correlation coefficients in the correlation matrix need be calculated before the ACELP search procedure is carried out. Furthermore, the proposed joint scheme, by combining the SCM method and a pulse position prediction scheme, not only decreases the arithmetic complexity in the pre-computing autocorrelation matrix but also reduces the number of pulse position combinations. The simulation and experimental results show that the proposed method provides an effective reduction in the computational load of the ACELP codebook search procedure with no discernible degradation of the speech quality.

Candidate scheme for fast ACELP search

The algebraic code excited linear prediction (ACELP) algorithm has been adopted by many speech coding standards, due to low complexity and high quality in its analysis-by-synthesis optimisation. For further computational complexity reduction, the authors propose a fast ACELP algorithm using a designed pilot function to predict the predetermined candidate pulses. With candidate pulses, it is possible to not only reduce the number of search loops but also to avoid the computation of unnecessary correlation functions. The proposed candidate position scheme can be applied to all ACELP coders such as the ITU G-723.1 and G.729 as well as the GSM enhanced full rate (EFR) speech coding standards. Simulation results show that the computational load can be reduced by about 50-80% with almost imperceptible degradation in performance.

ITU-T G.729 Annex A: reduced complexity 8 kb/s CS-ACELP codec for digital simultaneous voice and data

This article describes the ITU-T Recommendation G.729 Annex A (G.729A) for encoding speech signals at 8 kb/s with low complexity. G.729A is the standard speech coding algorithm for multimedia digital simultaneous voice and data (DSVD). G.729A is bitstream interoperable with G.729; that is, speech coded with G.729A can be decoded with G.729, and vice versa. Like G.729, it uses the conjugate-structure algebraic code excited linear prediction (CS-ACELP) algorithm with 10 ms frames. However, several algorithmic changes have been introduced which result in a 50 percent reduction in complexity. This article describes the algorithm introduced to achieve the low complexity goal while meeting the terms of reference. Subjective tests showed that the performance of G.729A is equivalent to both G.729 and G.726 at 32 kb/s in most operating conditions; however, it is slightly worse in the case of three tandems and in the presence of background noise. A breakdown of the complexities of both G.729 and G.729A is also given.

Performance of ITU-T G.729 8 kb/s CS-ACELP speech codec with nonvoice narrowband signals

A report is given on the results of a series of objective measurements conducted by COMSAT in a laboratory environment aimed at characterizing the narrowband performance of the ITU-T G.729 8 kb/s conjugate-structure algebraic code-excited linear prediction (CS-ACELP) speech coder. The test procedures followed ITU-T Recommendation G.720, "Characterization of Low-Rate Voice Coder Performance with Non-Voice Signals". It was concluded that the G.729 algorithm has excellent performance with narrowband signals in general (e.g., single tones and DTMF signals). As for Signaling System No. 5 (SS5) interregister signals, the G.729 CS-ACELP frequently failed to correctly identify SS5 digit 6 in a number of occurrences, using worst-case analysis equipment. This indicates that the SS5 performance of G.729 codecs in trunks where SS5 is used should be carefully measured before the network planner decides on its deployment. Great care should also be taken for tandem connections, since no test has been performed for these configurations.

A toll quality 8 kb/s speech codec for the personal communications system (PCS)

A toll quality speech codec at 8 kb/s suitable for the future personal communications system is presented. The codec is currently under standardization by the ITU-T (successor of CCITT) where the codec terms of reference were mainly determined considering PCS application. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP) and has a speech frame of 10 ms. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Formal subjective listening tests, performed by ITU-T SG 12, showed that the codec quality is equivalent to that of G.726 ADPCM at 32 kb/s in error-free conditions and it outperforms G.726 under error conditions. The codec performs adequately under tandeming conditions, and can support a frame erasure rate up to 3% with a degradation in its performance that is still worse than the ITU-T requirements, and this is one subject of study for the next phase. The algorithm has been implemented on a single fixed-point DSP for the ITU-T subjective rest, and required about 29 MIPS. An optimized version, however, requires 24 MIPS without any speech quality degradation.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>