Sharif Digital Repository

In this paper, a low order recursive linear prediction method and recursive least square as an adaptive filter (LP-RLS) are introduced to predict the speech and the excitation signals. In real-time packet-based communication systems, one major problem is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packet is known as lost packet. The easiest task of a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid quality reduction due to packet loss, a suitable method and/or algorithm is needed to replace the missing segments of speech. The... 

In real-time packet-based communication systems, one major problem is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packet is known as lost packet. The easiest task of a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid quality reduction due to packet loss, a suitable method and/or algorithm is needed to replace the missing segments of speech. In this paper, we introduce a low order recursive linear prediction method for replacement of lost speech segment. In this method a normalized least mean square (NLMS) as an adaptive filter... 

This paper describes a vector quantization of a 600 bps speech coding parameters based on the Mixed Excitation Linear Prediction (MELP) model, which was accepted as a standard in communication on narrow-band HF channels. The MELP speech coders are robust in difficult background noise environments and intended mostly for military communications. To reduce the bit rate, a joint vector quantization of multi-frame is developed that takes advantage of inherent inter-frame redundancy of the MELP parameters. By grouping parameters of 4 frames into a multi-frame and using vector quantization, bit rate is decreased 4 times and output speech is still intelligible  

This paper addresses a new mixed model for characterizing LPC excitation on a 3-band basis through analyzing harmonic structure of the residual signal. In addition, a sub-frame based analysis is developed for detecting both aperiodic pulses and noisy signals, which plays a major role in reduction of perceptual errors introduced by some certain consonants. Preliminary results show that near natural speech is achieved at 1050 bps, allocated to the excitation parameters, suggesting superiority of the proposed coding scheme to the MELP-2400 coding standard, in the sense of perceptual quality of reconstructed speech  

A major problem in real-time packet-based communication systems, is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packets are considered as lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the... 

A major problem in real-time packet-based communication systems, is misrouted or delayed packet which results in degraded perceived voice quality. If packets are not available on time, the packets are considered lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the clean... 

A major problem in real-time packet-based communication systems, is misrouted or delayed packets which results in degraded perceived voice quality. If packets are not available on time, the packets are considered as lost. The easiest solution in a network terminal receiver is to replace silence for the duration of lost speech segments. In a high quality communication system, to avoid degradation in speech quality due to packet loss, a suitable method or algorithm is needed to replace the missing segments of speech. In this paper, we introduce an adaptive filter for replacement of lost speech segment. In this method Kalman filter as a state-space based method will be used to predict the clean... 

This paper compares and quantifies the differences between formants of speech across accents. The crossentropy information measure is used to compare the differences between the formants of vowels of three major English accents British, American and Australian. An improved formant estimation method, based on a linear prediction model feature analysis and a hidden Markov model of formants, is employed for estimation of formant trajectories of vowels and diphthongs. The impact of vocal tract length on accent is also examined. Comparative analysis of formant space of the three accents indicates that these accents are mostly conveyed by the first two formants  

Adaptive Noise Cancellers (ANCs) do not provide sufficient noise reduction in the diffuse noise fields. In this paper, a new hybrid structure is proposed as a solution to this problem. The proposed system is a combination of two subsystems, an ANC and a new multistage post-filter. The post-filter is based on linear prediction (LP) and attempts to extract speech component by using intermediate ANC signals. The system is implemented on an over-sampled DFT filterbank with different analysis and synthesis prototype filters. The experimental results using various quality measures show that the proposed system is superior to both the subband ANC and subband LP based speech enhancement systems.1 ©... 

Using a proper distribution function for speech signal or for its representations is of crucial importance in statistical-based speech processing algorithms. Although the most commonly used probability density function (pdf) for speech signals is Gaussian, recent studies have shown the superiority of super-Gaussian pdfs. A large research effort has focused on the investigation of a univariate case of speech signal distribution; however, in this paper, we study the multivariate distributions of speech signal and its representations using the conventional distribution functions, e.g., multivariate Gaussian and multivariate Laplace, and the copula-based multivariate distributions as candidates.... 

Adaptive Noise Cancellation (ANC) is a well-known technique for background noise reduction in automobile and vehicular environments. The noise fields in automobile and other vehicle interior obey the diffuse noise field model closely. On the other hand, the ANC does not provide sufficient noise reduction in the diffuse noise fields. In this paper, a new multistage post-filter is designed for ANC as a solution to diffuse noise conditions. The designed post-filter is a single channel Linear Prediction (LP) based speech enhancement system. The LP is performed by an adaptive lattice filter and attempts to extract speech components by using intermediate ANC signals. The post-filter has no... 

Main objective of this article is to present the implementation and simulation of a Conjugate Structure Algebraic Code Excited Linear Prediction speech coder (CSACELP) based upon ITU-T's G.729 recommendation and to optimize it for real-time implementation on an FPGA. The suggested architecture is characterized by pipelining and parallel operation of functional units; using fixed point two's complement representation for integers. The design was functionally verified by utilizing the ModelSim software package from Mentor Graphics Corporation Company and then synthesized by Xilinx Integrated Software Environment (ISE) 6.1 software. Preliminary results show that the overall system delay is less...