Perceptual linear predictive noise modelling for sinusoid-plus-noise audio coding
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Sinusoidal coding of an audio subject to a bit-rate constraint, in general, results in a noise-like residual signal. This residual signal is of high perceptual importance; reconstruction of audio using the sinusoidal representation only typically results in an artificial sounding reconstruction. We present a new method, called perceptual linear predictive coding (PLPC), where the residual is encoded by applying LPC in the perceptual domain. This method minimizes a perceptual modelling error and therefore represents only residual components that are of perceptual relevance, while automatically discarding components masked by the sinusoidally coded part. Subjective listening tests show that PLPC performs significantly better than ordinary LPC as a sinusoidal residual coding technique. Furthermore, PLPC combined with a flexible segmentation and model order allocation algorithm leads to a significant gain in terms of R/D performance for fragments with fast changing characteristics.Generally, the code excited linear predictive (CELP) coding is known as one of the best algorithms for bit rate between 4 kbps and 16 kbps. Recently, the algebraic CELP (ACELP) algorithm has been widely adopted in standard speech coders such as G.729 and Adaptive multi rate (AMR) and AMR Wide Band (AMR-WB).In order to improve the speech quality based on CELP algorithm, harmonic Pitch pre-emphasis is employed on the impulses and a new codebook search method is proposed. Pitch pre-emphasis is applied in the analysis-by-synthesis loop of the analyzer, so the harmonic structure is strengthened. The new codebook search method reduces the computations considerably at the expense of a slight degradation of speech quality. The experiment result shows that the speech quality of integrated speech coding algorithm offers almost the same quality as G.729 while takes less search space.
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At low bit rates (around and below 4 kbit/s), the use of a rigid coding configuration in the CELP (Code Excited Linear Prediction) speech coding structure leads to a bad reconstruction of the voiced onsets. In this paper we describe a 4 kbit/s CELP algorithm that gives priority to these critical speech segments. Our scheme employs different combinations of codebooks and bit allocations for different speech classes. Mean opinion scores (MOS) reveal that at 4 kbit/s the speech quality obtained with our coding algorithm is significantly better than that achieved with a CELP codec that does not use different coding strategies for different classes of speech.
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A novel variable-rate linear predictive coding (LPC) parameter quantization scheme is proposed, in which linear prediction is done by using either the current (forward LPC) or previously decoded (backward LPC) speech blocks. The proposed LPC quantization scheme was integrated into the FS1016 Federal Standard CELP coder. Significant LPC bit rate reduction is achieved without compromising the decoded speech quality.
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In this paper,we analyse the new CELP (code excited linear predictive)coder in audio part of MEPG4.A speech coding prototype system based on CELP for narrow band speech is designed and implemented.An idea of utilizing the CELP speech coding technique for compression of speech database in TTS (text to speech) conversion system is proposed.Synthesis experiments show the decoded speech is high natural.
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In recent years much work has be concentrated on encoding speech. Various coding methods are developed such as Low-Delay Codebook Excited Linear Prediction (LD-CELP), Codebook
Excited Linear Prediction (CELP), Pulse Residual Excited Linear Prediction (PRELP) and others. These encoders operate at very low bit rates (4.8Kbps) as well as maintained speech quality. In this chapter some ofthese encoding algorithms are explained
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This paper presents an algorithm for fast codebook search of code excited linear prediction (CELP) coders and its descendants. The problem of reducing the bit rate of speech while preserving the quality of speech reconstructed from such a representation has received continuous attention. Real time implementation of adaptive codebook search in code excited linear prediction (CELP) and CELP based speech coders is identified as the computationally most complex module. Thus in this work, we propose a new method of using one bit computation instead of the 16 bit computation in the codebook search part of the speech codec. The simulation results show that effective codebook size and hence the computational time is reduced by 50%.
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A new speech coding technique at low bit-rate is presented in this paper. The coder is based upon a novel speech production model, independently developed by the authors [1,2] and by Atal and Schroeder [3,4], called CELP (Codebook Excited Linear Prediction). Differences exist between the two approaches, both in the strategy chosen to construct codebooks, and in the method to generate the innovation sequence. In this scheme, we split the incoming speech signal into two frequency bands in order to gain the benefits of the piecewise LP (Linear Prediction) approximation. Then, each residual signal is coded in blocks of 5-ms duration through an adaptive vector quantizer incorporating a noise shaping filter. Our results show that good quality speech can be obtained at 8 kbit/s.
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We present an approach to low-rate speech coding based on multistage self-excited linear prediction (SELP). The coder has much lower complexity than a code-excited linear prediction (CELP) coder, but yields comparable output speech quality at the same transmission rate. In the proposed multistage SELP coder, the long-term predictor is modified to model better the onset of a voiced section of speech or voice with high pitch.
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The last few years has seen a rapid expansion in the development of efficient speech compression algorithms which has been primarily fuelled by the proliferation of digital mobile communication systems. Low bit rate speech coding algorithms estimate, quantise and efficiently encode the parameters of a speech production model by using the original speech waveform. The most popular of these models is based on the technique of Linear Prediction which has resulted in a class of speech coding algorithms known as Analysis-by-Synthesis Linear Prediction Coding (AbS-LPC). In the AbS-LPC coding system, a closed loop optimisation procedure is used to determine the excitation signal for the Linear Prediction filter. This methodology of speech coding has been the foundation of many algorithms operating at medium to low bit rates. In particular, the Codebook Excited Linear Prediction (CELP) algorithm has received much attention in the past few years which has culminated in numerous standards being based on this principle. CELP achieves its coding efficiency and high quality by representing the excitation signal as a vector. However, in the original implementation of this algorithm the excitation search was very computationally intensive due to the structure of the codebook. In order to reduce this computational complexity and improve the quality of the synthetic speech this thesis explores various structures of secondary excitations which are based on sparsely populated pulsed vectors. A variable rate implementation of the CELP algorithm is also presented where techniques typically found in vocoders are used to provide an accurate classification of the different types of speech. These metrics are then used to vary the speech segment size and coding rate to take advantage of the differing regions of speech. Narrowband speech is defined to be band limited between 300 Hz - 3.4 kHz and is sampled at the Nyquist sampling rate of 8 kHz. However, wideband speech lies between 50 Hz and 7 kHz and is consequently sampled at a higher rate of 16 kHz. Wideband speech exhibits characteristics which are not normally embodied within the narrowband signal. It is these characteristics which contribute to the superior perceived quality and therefore it is imperative that a coding scheme maintains this information. This thesis formulates various strategies for the coding of wideband speech using the CELP coding structure. Particular attention is paid to preserving the information in the higher frequencies so that the overall quality is maintained in the synthetic signal. A low delay variant of the wideband coder is also presented where particular attention is paid to the effects of backward LPC prediction over the full bandwidth of the signal are investigated. This results in a split band architecture which is capable of producing high quality wideband speech.
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Perceptual linear prediction (PLP) is widely used in speech recognition systems as a feature extraction method. Also code-exited linear prediction coder (CELP) is one of the well known speech coders which widely used in communication systems. In this paper the application of PLP in speech coding has been discussed. In the first stage the parameters of formant synthesis filter are determined by applying PLP algorithms. Then these parameters are used in coder, code-exited linear prediction coder, to improve the efficiency of this kind of coder. The experiments show promising result in some cases.
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