CODE EXCITED LINEAR PREDICTION FOR SPEECH COMPRESSION

2000

This paper discusses the design and implementation of a low bit rate codec along with its performance at different bit rates. The International Telecommunications Union's G.728 CELP speech coder is specifically designed for low coding delay and toll quality speech at a rate of 16kbps. Here, we present the design of a CELP(Code Excited Linear Prediction) algorithm similar to the above coder, but can compress speech up to rates like 6.4kbps(as in the G.723.1 standard). This codec shares the advantages of both the above standards; it has a low delay (up to 1.25ms) and greater compression, even though the techniques used are simpler. This advantage makes this coder suitable for applications like video conferencing, mobile communications etc.

Electronics and Communications in Japan (Part III: Fundamental Electronic Science), 1994

This paper proposes a new speech coding method pitch synchronous innovation code excited linear predictor (PSI-CELP). This method is based on CELP but adds pitch synchronous innovation. This results in even random codevectors being adaptively converted to have pitch periodicity for voiced frames. This scheme can improve the synthesized speech quality of voiced frames in the low bit-rate CELP without increasing either computational complexity or bit rate.

1999

This paper presents a novel multi-band CELP coder with the following characteristics: wideband coding (6.5 kHz), variable bit rate (VBR) coding (10-24 kbps), low-delay (10 ms), embeddibility, and perceptually based dynamic bit allocation. The excitation signal of the linear prediction filter is the vector sum of eight off-line pre-filtered bandpass excitation vectors. The eight excitation codebooks are tree structured, providing embeddibility and variable bit rate. The dynamic allocation of the bitstream among the different bands is based on the perceptual importance of each band. The multi-band and perceptual structure of the coding scheme results in graceful degradation with decreasing bit rates both in quiet and in the presence of background noise.

In this paper we study the performance and the error sensitivities of six CELP [I] based codecs operating between 8 and 4 kbitsls. Codecs using both forward and backward adaption of the linear prediction coefficients and the long term predictor (LTP) are described. Initially we describe four low delay codecs which all use backward adaption of the LPC coefficients but which differ in their use of LTP. These codecs all have frame-lengths of 3 ms or less, and their performance at various bit rates between 8 and 4 kbitafs is examined. Next the error sensitivity of these codecs, and means of improving it, are described. Then an algebraic CELP (ACELP) [2] codec operating at 6.2 kbitsfs with a frame-length of 5 ms is described. Our final codec also uses ACELP and operates between 4.7 and 7.1 kbitsls, but it is forward adaptive and so it has a much longer frame-length of up to 30 ms. After describing this codec we compare the performance of our codecs in both error-he conditions and in the presence of channel errors. Surprisingly the error sensitivity of the low delay backward adaptive codec with no Lll' is similar to that of the forward adaptive, high delay,ACELP codec.

International Journal of Computer Applications, 2012

The critical issues that are serving as constraints in wireless communication particularly in mobile communication are bandwidth, storage memory and power. The speech transmission in wireless networks is associated with the reduction of extra information present in signal in such a way to preserve the quality and intelligibility of speech. To remove the redundancy and transmit the speech with acceptable quality, speech compression algorithms are deployed. Because of this reason the speech coding is and will be the most important research issue. This paper addresses the implementation of CELP coder having low computational complexity with acceptable speech quality and preserves the intelligibility. The coder is assessed in terms of quality for different kinds of speakers using PESQ, PSNR,Frequency Weighted SNRseg, and SNRseg.

1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258), 1999

CELP coders using pulse codebooks for excitations such as ACELP[1] have the advantages of low complexity and high speech q u a l i t y. At l o w bit rates, however, the decrease of pulse position candidates and the numberof pulses degrades reconstructed speech quality. This paper describes a method for adaptive allocating of pulse position candidates. In the proposed method, N ecient candidates of pulse positions are selected out of all possible positions in a subframe. Amplitude envelope of an adaptive code vector is used for selecting N ecient candidates. The larger the amplitude is, the more pulse positions are assigned. Using an adaptive c o d e v ector for the adaptation, the proposed method requires no additional bits for the adaptation. Experimental results show that the proposed method increases WSNRseg by 0.3dB and MOS by 0.15.

European Transactions on Telecommunications, 2010

A variable bit-rate speech coder intended for digital cellular applications is described. A voice activity detection algorithm is used to distinguish active speech from background noise. Each frame of active speech is further classified to distinguish between three phonetic categories: voiced, unvoiced, and onset. Each input frame is assigned one of five bit rates according to voice activity and phonetic classification and coded using an analysis-by-synthesis algorithm tailored to the needs of the class that it belongs to. The resulting coder, called Variable Rate Phonetic Segmentation, produces good quality speech at an average bit-rate below 3 kbit/s when operating with a voice activity factor of 0.5. Informal subjective quality assessment for speech in clean and noisy backgrounds indicates a performance that is comparable to the TIA standard QCELP algorithm while operating at a 25% to 40% lower average bit rate. (') This work was supported by Fujitsu Laboratories Ltd., the National Science Foundation, the UC Micro program, Rcckwell International Corporation. Hughes Aircraft Company, and Qualcomm, Inc. (2) K. Srinivasan is currently with Advanced Comprrssion Technology. Inc.. Simi Valley, CA.

2017 25th European Signal Processing Conference (EUSIPCO), 2017

The Code-Excited Linear Prediction (CELP) model is very efficient in coding speech at low bit rates. However, if the bit rate of the coder is increased, the CELP model does not gain in quality as quickly as other approaches. Moreover, the computational complexity of the CELP model generally increases significantly at higher bit rates. In this paper we focus on a technique that aims to overcome these limitations by means of a special transform-domain codebook within the CELP model. We show by the example of the AMR-WB codec that the CELP model with the new flexible and scalable codebook improves the quality at high bit rates at no additional complexity cost.

International Conference on Acoustics, Speech, and Signal Processing, 1993

A high-quality 8-bit/s speech coder based on CS:CELP (conjugate structure code excited linear prediction) with 10 ms frame length is presented. To provide high quality in both error-free and error conditions, it uses four schemes: LSP (line spectrum pair) quantization using interframe correlation, preselection of codebook search, a conjugate structure, and backward adaptation of the VQ (vector quantization) gain. LSP

Electronics and Communications in Japan (Part III: Fundamental Electronic Science), 2000

This paper presents a variable bit rate ADP-CELP (Adaptive Density Pulse Code Excited Linear Prediction) coder that selects one of four kinds of coding structure in each frame based on short time speech characteristics. To improve speech quality and reduce the average bit rate, we have developed a speech/non-speech classification method using spectrum envelope variation, which is robust for background noise. In addition, we propose an efficient pitch lag coding technique. The technique interpolates consecutive frame pitch lags and quantizes a vector of relative pitch lags consisting of variation between an estimated pitch lag and a target pitch lag in plural subframes. The average bit rate of the proposed coder was approximately 2.4 kbps for speech sources with activity factor of 60%. Our subjective testing indicates the quality of the propcsed coder exceeds that of the Japanese digital cellular standard with rate of 3.45 kbps.