Code Excited Linear Prediction Coder Research Articles

Low Delay Sparse and Mixed Excitation CELP Coders for Wideband Speech Coding

Code Excited Linear Prediction (CELP) algorithms are proposed for compression of speech in 8 kHz band at switched or variable bit rate and algorithmic delay not exceeding 2 msec. Two structures of Low-Delay CELP coders are analyzed: Low-delay sparse excitation and mixed excitation CELP. Sparse excitation is based on MP-MLQ and multilayer models. Mixed excitation CELP algorithm stems from the narrowband G.728 standard. As opposed to G.728 LD-CELP coder, mixed excitation codebook consists of pseudorandom vectors and sequences obtained with Long-Term Prediction (LTP). Variable rate coding consists in maximizing vector dimension while keeping the required speech quality. Good speech quality (MOS=3.9 according to PESQ algorithm) is obtained at average bit rate 33.5 kbit/sec.

A Low-Complexity Search Alagorithm for Speech Coders with Sparse Excitation 10.14209/jcis.2001.15

A low-complexity joint position and amplitude search algorithm is proposed for deterministic sparse codebooks to be used in code-excited linear predictive coders. The conventional multipulse search is presented as an extension of a general two-stage sequential search, leading to a classification of multistage sequential searches according to the extent that orthogonality constraints are applied throughout the search stages. After its classification, an efficient implementation of the joint search is derived which incorporates backward filtering of the residual target vector and precornputation of autocorrelation elements, bringing about a reduction in complexity of approximately one third in comparison to the focused search within the G.723.1 codec. In particular, considering an extensive speech database, a floatingpoint version of the joint search method reduces down to one third the number of comparisons per subframe relative to the focused search. Moreover, the joint search performs about one thirtieth as many comparisons as the position-exhaustive search. Further, the complexity of a fixed-point implementation of the joint search is below one quarter that of the focused search and stands below those of the G.729A and lS-64l-A coders. Listening tests indicate an equivalence in perceived quality.

Scalable Speech Coding for IP Networks: Beyond iLBC

High quality speech at low bit rates makes code excited linear prediction (CELP) the dominant choice for a narrowband coding technique despite the susceptibility to packet loss. One of the few techniques which received attention after the introduction of CELP coding technique is the internet low bitrate codec (iLBC) because of inherent high robustness to packet loss. Addition of rate flexibility and scalability makes the iLBC an attractive choice for voice communication over IP networks. In this paper, performance improvement schemes of multi-rate iLBC and its scalable structure are proposed, and the proposed codec enhanced from the previous work is re-designed based on the subjective listening quality instead of the objective quality. In particular, perceptual weighting and the modified discrete cosine transform (MDCT) with short overlap in weighted signal domain are employed along with the improved packet loss concealment (PLC) algorithm. The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions. This result is significant considering that development of the proposed codec is still in early stage.

Overall performance evaluation of adaptive multi rate 06.90 speech codec based on code excited linear prediction algorithm using MATLAB

Today, the primary constrain in wireless communication system is limited bandwidth and power. Wireless systems involved in transmission of speech envisage that efficient and effective methods need to be developed for maintaining quality-of-speech, especially at the receiving end, with maximum saving of bandwidth and power. Amongst all elements of the communication system (transmitter, channel and receiver), transmission channel (carrier of information/data, also called the medium) is the most critical and plays a key role in the transmission and reception of information/data. Channel conditions decide the quality of speech at receiver. Modeling a channel is a complex task. Many techniques are adopted to mitigate the effect of the channel. AMR (Adaptive Multi Rate) is one such technique that counteracts the deleterious effect of the channel on speech. This technique employs variable bit rate that dynamically switches to specific modes of operation (switching bit rates--called modes of operation) depending upon the channel conditions. In this paper, the application of Code Excited Linear Prediction (CELP) source coder on speech followed by AMR codec is investigated and studied. An e-test bench using MATLAB is created to implement the CELP based AMR Codec scheme, and the same studied and investigated through a series of simulation. Here, both subjective and objective evaluations are carried out. Objective evaluations are categorized into waveform based, spectral based and perceptual based analysis. The results of the simulations are recorded and compared in various graphs and tables, which include calculation of various parameters like Absolute Error (ABS), Mean Square Error (MSE), Root Mean Square Error (RMSE), Signal to Noise Ratio (SNR), segmental SNR (segSNR) (Y. Hu and P. Loizou in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., vol. 1, pp. 153---156, 2006a; Proc. Interspeech, pp. 1447---1450, 2006b), Weighted-Slope Spectral distance (WSS) (Y. Hu and P. Loizou in Speech Commun. 49, 588---601, 2007), Perceptual Evaluation of Speech Quality (PESQ) (ITU-T rec. P.862, 2000), Log-Likelihood Ratio (LLR), Itakura-Saito Distance measure (ISD), Cepstrum Distance Measures (CEP) (V. Turbin and N. Faucheur in Proc. Online Workshop Meas. Speech Audio Quality Netw., pp. 81---84, 2005), Frequency Weighted Segmental SNR (fwSNRseg), Predicted rating of overall Quality (Covl), Rating of Speech Distortion (Csig), Rating of Background Distortion (Cbak) (ITU-T rec. P.835, 2003) and MeanOpinion Score (MOS). Simulation results clearly advocate that, it is possible to producevariable bitrates (tuning to channel conditions) in CELP coder by affecting coefficients of the coder while still maintaining a good quality of speech. Further, higher the bit-rate used, the better is the quality of speech (which can be verified from the results obtained with PESQ and MOS analysis) and at the same time offered simulation delay time also increases.

Soft Reconstruction of Speech in the Presence of Noise and Packet Loss

Exploiting the residual redundancy in a source coder output stream during the decoding process has been proven to be a bandwidth efficient way to combat the noisy channel degradations. In this paper, we consider soft reconstruction of speech spectrum, in GSM adaptive multirate and IS-641 vocoders, transmitted over a channel disturbed with noise and/or packet loss. Several schemes are presented which exploit different levels of intraframe and interframe residual redundancy for improved source decoding at the receiver. A packetization strategy is proposed which is matched to the presented error concealment units. For decoders that exploit the residual redundancy, extensive complexity has been a serious concern, especially as the quantizer bitrate increases . In this paper, a novel method is presented to construct reduced complexity algorithms. The proposed methodology is based on the classification of the signal domain and efficient approximation of the residual redundancy or the a priori transition probabilities. The presented schemes provide high quality error concealment solutions for code excited linear prediction (CELP) coders

The history of linear prediction

This paper recollects the events that led to proposing the linear prediction coding (LPC) method, then the multipulse LPC and the code-excited LPC

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.

Multiband Vector Quantization Based on Inner Product for Wideband Speech Coding

This paper describes a multiband vector quantization (VQ) technique based on inner product for wideband speech coding at 16 kb/s. Our approach consists of splitting the input speech into two separate bands and then applying an independent coding scheme for each band. A code excited linear prediction (CELP) coder is used in the lower band while a transform based coding strategy is applied in the higher band. The spectral components in the higher frequency band are represented by a set of modulated lapped transform (MLT) coefficients. The higher frequency band is divided into three subbands, and the MLT coefficients construct a vector for each subband. Specifically, for the VQ of these vectors, an inner product-based distance measure is proposed as a new strategy. The proposed 16 kb/s coder with the inner-product based distortion measure achieves better performance than the 48 kb/s ITU-T G.722 in subjective quality tests.

Voice Activity Detection Algorithm Based on Radial Basis Function Network

This paper proposes a Voice Activity Detection (VAD) algorithm using Radial Basis Function (RBF) network. The k-means clustering and Least Mean Square (LMS) algorithm are used to update the RBF network to the underlying speech condition. The inputs for RBF are the three parameters a Code Excited Linear Prediction (CELP) coder, which works stably under various background noise levels. Adaptive hangover threshold applies in RBF-VAD for reducing error, because threshold value has trade off effect in VAD decision. The experimental results show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

Watermarking Combined with CELP Speech Coding for Authentication

This letter presents a speech watermarking scheme that is combined with CELP (Code Excited Linear Prediction) speech coding for speech authentication. The excitation codebook of CELP is partitioned into three parts and labeled '0', '1' and 'any' according to the private key. Watermark embedding process chooses the codebook whose label is the same as the watermark bit and combines it with the codebook labeled 'any' for CELP coding. A statistical method is employed to detect the watermark, and the watermark length for authentication and detection threshold are determined by false alarm probability and missed detection probability. The new codebook partition technique produces less distortion, and the statistical detection method guarantees that the error probability can be controlledunder prescribed level.

KLT-Based Adaptive Classified VQ of the Speech Signal

Compared to scalar quantization (SQ), vector quantization (VQ) has memory, space-filling, and shape advantages. If the signal statistics are known, direct vector quantization (DVQ) according to these statistics provides the highest coding efficiency, but requires unmanageable storage requirements if the statistics are time varying. In code-excited linear predictive (CELP) coding, a single compromise codebook is trained in the excitation-domain and the space-filling and shape advantages of VQ are utilized in a nonoptimal, average sense. In this paper, we propose Karhunen-Loe/spl grave/ve transform (KLT)-based adaptive classified VQ (CVQ), where the space-filling advantage can be utilized since the Voronoi-region shape is not affected by the KLT. The memory and shape advantages can be also used, since each codebook is designed based on a narrow class of KLT-domain statistics. We further improve basic KLT-CVQ with companding. The companding utilizes the shape advantage of VQ more efficiently. Our experiments show that KLT-CVQ provides a higher SNR than basic CELP coding, and has a computational complexity similar to DVQ and much lower than CELP. With companding, even single-class KLT-CVQ outperforms CELP, both in terms of SNR and codebook search complexity.

Voice activity detection algorithm using radial basis function network

A voice activity detection (VAD) algorithm using the radial basis function (RBF) network is proposed. The k-means clustering and least mean square algorithms are used to update the RBF network to the underlying speech condition. The inputs for RBF are code excited linear prediction coder parameters, which are stable under background noise. The RBF network output is compared to a threshold to determine the nature of the period (voice or silence). Experimental results show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

Development of a 4 kbit/s hybrid sinusoidal/CELP speech coder

A comprehensive performance analysis of sinusoidal and code excited linear prediction (CELP) speech coding is given around 4 kbit/s, using both subjective and objective measurements. Based on the observations made, justification for the multi-modal hybrid coding approach employing both sinusoidal and CELP coding is given, and an implementation of such a coder is described. This 4 kbit/s sinusoidal/CELP speech coder utilizes four modes to classify the input speech segment: voiced, jittery-voiced, plosive and unvoiced. For voiced segments sinusoidal coding is used whereas different CELP versions are employed for the other modes. The quality of the implemented 4 kbit/s sinusoidal/CELP speech coder in clean speech conditions is finally verified by a listening test. In the test, the 4 kbit/s coder performed almost as well as the high-quality references used, but it still needs improvements to be classified as a high-quality 4 kbit/s speech coder.

Trellis code excited linear prediction (TCELP) speech coding

This paper describes extensions to trellis-based scalar-vector quantization (TB-SVQ) , as novel techniques for coding sources with memory. These techniques are proposed as efficient solutions for the quantization of the excitation in Code-Excited Linear Predictive (CELP) coding algorithms. A new 24 kbit/s low-delay trellis CELP (LD-TCELP) coder is introduced which achieves high-performance coding of narrowband speech with an algorithmic delay of only 3 ms. This coder utilizes the granular structure of the underlying trellis code and the codebook shaping capability of the SVQ approach to effectively capitalize on the gain of higher dimensional (24-D) vector quantization. We present experimental results which show that the LD-TCELP coder outperforms two toll-quality standardized coders: ITU-T G.728 16 kbit/s LD-CELP and ITU-T G.726 32 kbit/s ADPCM. The proposed LD-TCELP coder can also utilize a low-complexity codebook search approach with a computational complexity that is about 75% of that of ITU-T G.728 LD-CELP.

A new discrete spectral modeling method and an application to CELP coding

In this letter, a new discrete spectral modeling method is proposed based on a comparison of distance measure performance in an adaptive filtering context. It is shown that a new fast converging adaptive algorithm yields a more accurate estimate of the spectral envelope of the speech spectrum by minimizing COSH distance rather than the Itakura-Saito distance measure. We apply discrete spectral all-pole modeling to code-excited linear predictive coding by refining the short-term synthesis filter coefficients originally obtained by the linear prediction method. Simulation results show the enhanced harmonic and formant structure in the speech spectrum and that better speech quality is obtained.

On Rationally DSP Implementation of the MP-MLQ/ACELP Dual Rate Speech Encoder for Multimedia Communications

The excellent performance in communications quality speech coding below 8 kbps achievable with the code-excited linear prediction (CELP) coders gives to this architecture a predominant role in medium-rate and low-rate speech coding, as evidenced by the adoption of several recent fixed-rate and variable-rate standards. Unfortunately, some of these CELP-based schemes are not completely described in the literature, and consequently they are difficult to understand and implement efficiently. This paper presents an original study of the G723.1 codec. The G723.1 encoder is dedicated to compress the voice signals with bandwidth up to 4 kHz efficiently and to deliver an encoded data stream with a very low binary rate and a good quality of transmitted speech (typical applications being encoding of the vocal signal for video conferences via GSTN and Voice over IP). We perform a detailed and gradually analysis, describing the MP-MLQ/ACELP speech coder from the point of view of a classical CELP structure. This approach allows us to identify (using theoretical considerations) the starting internal structure of each processing block from the encoder scheme. These results are used in breaking the main encoding algorithm loop. Finally, using the previously revealed starting internal structure, we derive the algorithm for the pitch predictor block, which is one of the most difficult parts of the ITU-T G723.1 encoder. The accompanying comments, explanations and diagrams allow efficient implementation and debugging of the corresponding software by regular DSP programmers.

A Low-Complexity Search Alagorithm for Speech Coders with Sparse Excitation

A low-complexity joint position and amplitude search algorithm is proposed for deterministic sparse codebooks to be used in code-excited linear predictive coders. The conventional multipulse search is presented as an extension of a general two-stage sequential search, leading to a classification of multistage sequential searches according to the extent that orthogonality constraints are applied throughout the search stages. After its classification, an efficient implementation of the joint search is derived which incorporates backward filtering of the residual target vector and precornputation of autocorrelation elements, bringing about a reduction in complexity of approximately one third in comparison to the focused search within the G.723.1 codec. In particular, considering an extensive speech database, a floatingpoint version of the joint search method reduces down to one third the number of comparisons per subframe relative to the focused search. Moreover, the joint search performs about one thirtieth as many comparisons as the position-exhaustive search. Further, the complexity of a fixed-point implementation of the joint search is below one quarter that of the focused search and stands below those of the G.729A and lS-64l-A coders. Listening tests indicate an equivalence in perceived quality.

A frame erasure concealment algorithm based on gain parameter re-estimation for CELP coders

In this paper, we propose a frame erasure concealment algorithm based on reestimating gain parameters for a code-excited linear prediction (CELP) coder. When a frame is detected as being erased, the coding parameters, especially the adaptive codebook gain and fixed codebook gain, of the erased and subsequent frames, are reestimated by a gain-matching procedure. By doing this, we can reduce the abrupt change caused in the decoded excitation signal by a simple scaling down procedure. We have applied this technique to the IS-63-1 speech coder and found that the proposed algorithm improves the speech quality under various channel conditions compared with the conventional extrapolation-based concealment algorithm.

Joint position and amplitude search of algebraic multipulses

A joint position and amplitude search algorithm is proposed for algebraic multipulse codebooks to be used in code-excited linear predictive (CELP) coders. The joint search complexity is below one quarter that of the focused search and ranks below those of the G.729A and IS-641-A coders. Listening tests indicate an equivalence in perceived quality.

Pitch synchronous extended excitation in multimode CELP

This letter proposes a 4-kb/s multimode code-excited linear prediction (CELP) coder with pitch synchronous extended excitation. Three modes are used for the short-term excitation, namely algebraic, extended, or stochastic excitations, together with an adaptive codebook for the long-term excitation. Comparisons with the FS-1016 and ITU-T G.723.1 coders show a performance level between these standards.