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.

Variable-rate speech coding: Replacing unvoiced excitations by linear prediction residues of different phonemes

Abstract: In order to reduce the bit rate of speech transmission while maintaining the speech quality we are developing a vocoder which uses different methods for coding voiced and unvoiced frames. Within this framework we present an idea for expressing fricative and plosive phonemes with only 20 bits per frame (tD 20ms). We show that they can be represented by LP(Linear Prediction) coefficients and a residual signal where this residue is always taken from a fixed phoneme of a test speaker known at the receiver station of the coding system (see figure 1). Algorithms ensuring smooth transitions to other speech-frame categories are also described below. Using this technique the transmission rate of unvoiced frames can be considerably reduced (down to 1 kbit/s) getting better listening results than using CELP (Code Excited Linear Prediction) variants at 4 kbit/s instead. The resulting ‘Multi-Class Vocoder’ (voiced frames are coded by Harmonic Coding at 4 kbit/s) has a variable rate of less than 3 kbit/s on the average.

Combining parametric and waveform-matching coders for low bit-rate speech coding

Abstract: Waveform-matching coders preserve the shape of the target waveform and the time synchrony between the original and the synthesized signal In parametric coders the shape of the encoded waveform is often changed, and the time-synchrony between the input and synthesized speech is not preserved These two issues, time synchrony and waveform shape, are major obstacles in representing different speech regions with parametric and waveform coders, as arbitrary switching between the two results in annoying artifacts in transition regions We describe a hybrid parametric/waveform coder with MELP used for strongly voiced regions and CELP employed for weakly voiced and unvoiced speech segments To limit switching artifacts between the coders, alignment phase is estimated and transmitted in MELP making the original and synthesized speech time-synchronous Additionally, in zero-phase equalization, the phase component of the CELP target signal is removed making the target waveform more similar to the MELP-synthesized speech These two techniques, alignment-phase encoding and zero-phase equalization, greatly reduce switching artifacts in transition regions between the parametric and waveform coders Formal listening tests of the 4 kb/s hybrid coder show that it can achieve speech quality equivalent to 32 kb/s ADPCM

Variable bit-rate celp coding of speech with phonetic classification

Abstract: A speech coding method using analysis-by-synthesis includes sampling an input speech and dividing the resulting speech samples into frames and subframes. The frames are analyzed to determine coefficients for the synthesis filter (136). The subframes are categorized into unvoiced (116), voiced (118) and onset (114) categories. Based on the category, a different coding scheme is used. The coded speech is fed into the synthesis filter (136), the output (138) of which is compared to the input speech samples (104) to produce an error signal (144). The coding is then adjusted per the error signal.

Linear Prediction Analysis and Quantization for the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction Speech Compression Algorithm

Abstract: Voice-over Internet Protocol (VoIP) telephony which is the transmission of real-time voice over an Internet Protocol (IP) data network, is increasingly becoming a variant telecommunication technology that one day may surpass the old analog and digital telephone systems. The Quality-of-Service (QoS) factor is an important parameter to be considered when measuring the performance of a VoIP system. Algorithmic delay (latency) may influence the QoS of a VoIP system. This paper presents the hardware implementation of the Linear Prediction (LP) Analysis and Quantization component of the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) speech compression algorithms. The LP Analysis and Quantization units were implemented in hardware using Xilinx 11.1 ISE, after which ITU-T test vectors were used to determine whether they were equivalent implementations of the LP Analysis and Quantization. The algorithmic delays of hardware-implementations were obtained via simulation using Modelsim XE 6.4b while the algorithmic delays of the software-implemented speech compression algorithms were obtained by software profiling using the GPROF profiler. After comparison the algorithmic delays of the hardware-implementations of the LP Analysis and Quantization was found to be shorter than the algorithmic delays of their software-implemented and hardware-implemented counterparts.