Code-excited linear prediction

About: Code-excited linear prediction is a research topic. Over the lifetime, 2025 publications have been published within this topic receiving 28633 citations. The topic is also known as: CELP.

Implementation of a 5.0 kb/s. coder for vehicular applications

Abstract: In thispaper andits companion, wepresent the results of our reseurch on the real-time implementation of a speech coding system for use in an automobile environment subject to both r4 engine, and wind noises and acoustic echos. In Part I, we will be emphasizing a TMS320C30-based implementation of an Finite-State VQ CELP Coder operating at 5,000 bits per second In Part I& we will be presenting, a TMS32OC50 implementation of an acoustic echo and rodengine noise canceller for trehicular applications designed to work with full-rate or harf-rate digital speech w&rs, including V.P, CELe FSVQ CUP, WC, QCELP and other derivatives of the CELP coder.

Lowdelay combined wideband speech and audio coding with scalable bit rates

Abstract: The integration of fixed and wireless networks have expanded the range in which speech and audio coders were designed to operate. As a result, current research is focusing on a number of developments including algorithms which are able to adapt to different transmission environments and to operate under multiple constraints of bit rate, complexity, delay, robustness to bit errors and diversity of input signals. In this paper, we propose a single coding algorithm for compressing wideband speech and audio signals (0-8kHz) operating with scalable bit rates and with low delay. The algorithm is based on the backward-adaptive linear predictive coding (BA LPC) technique in conjunction with an efficient closedloop optimised excitation structure consisting of sparse pulses of ternary values. The output bit rates range from 17 to 68kb/s. The scalability feature is achieved by means of discrete quantisation layers representing various levels of enhancements of the base-line coder and also flexibility in terms of complexity and bit allocation requirements depending on the particular application and on the network resources. An evaluation of the performance of the coder operating at 17kb/s is carried out using the G.722 standard as a reference.

A Silence Compression Algorithm for the Multi-Rate Dual-Bandwidth MPEG-4 CELP Standard

Abstract: This paper proposes a silence compression algorithm operating at multi-rates (MR) and with dual-bandwidths (DB), a narrowband and a wideband, for the MPEG-4 CELP standard. The MR/DB operations are implemented by a variable-frame-size/dual-bandwidth voice activity detection (VF/DB-VAD) module with domain conversions of the input signal, and a variable frame-size comfort noise generator (VF-CNG). The CNG module adaptively controls the power smoothing of the decoded signal to improve the coding quality. The algorithm also employs a dual-rate discontinuous transmission (DR-DTX) module to reduce an average transmission bit rate during silence periods. The subjective test results show that the proposed silence compression algorithm gives no degradation in coding quality for clean and noisy speech signals. These signals include about 20 to 30% nonspeech frames and the average transmission bit rates are reduced by 20 to 40%. The proposed algorithm has been standardized as a part of the ISO/IEC MPEG-4 CELP version 2.

Speech Compression Using Cosine Packet Decomposition.

Abstract: : As digitization of data becomes more prevalent, the demands on existing communications networks and computer systems to cope with this increase in data become overwhelming. Currently, the speech compression problem is handled using the CELP( Code Excited Linear Prediction) scheme and its derivatives. Such techniques are the most frequently used for speech compression at medium-to-low rate ranges. Recent research conducted into the area of cosine packets has proven this field to be readily adaptable to speech compression and coding. In this thesis, speech compression schemes are developed using cosine-packet decomposition, minimum entropy basis selection, and an adaptive thresholding scheme for selecting coefficients. In addition, voiced-unvoiced segmentation and a denoising scheme were implemented. Test results showed high compression ratios (1:50) with a good quality of reconstructed speech.

Analysis of LSF switched-predictive vector quantisers

Abstract: Switched-predictive vector quantisation (SPVQ) has proven to be more efficient than conventional predictive vector quantisation (PVQ) to reduce the bit rate of LSF parameters in speech coding applications. We perform a comparison between different SPVQ schemes and we propose an extension of the SPVQ technique which outperforms previously reported PVQ and SPVQ systems. The proposed method consists of 4 vector quantisers (VQ) which are switched on the basis of the best performance criterion.