A Hybrid Coding with Adaptive Filters Using Non-uniform Technique in Speech Signals
نویسندگان
چکیده
This paper proposes a new hybrid method in non-uniform sampling coding that uses adaptive filters as like zero-crossing and correlation. In speech signals, the modified zero-crossing rate method refers to the number of times in which the signal crosses zero in frame, the rate is high in noisy parts but low in voiced signal or unvoiced signal. And the correlation is high in voiced signals, low in noise. Also the non-uniform sampling uses the peaks and valleys, we consider that peaks and valleys is high in correlation. The method proposed in this paper uses the number of peaks and valleys in determining the threshold values of voiced, unvoiced signal and noise. Therefore, by decreasing the number of samples from sections with noise and voiceless sounds which have a high number of peaks and valleys, and by creating a signal that has little effect on the recognized signals, we may get higher compression rates. KeywordWaveform coding, non-uniform Sampling, Peak and Valley, Adaptive filter, Zero-crossing
منابع مشابه
A New Shuffled Sub-swarm Particle Swarm Optimization Algorithm for Speech Enhancement
In this paper, we propose a novel algorithm to enhance the noisy speech in the framework of dual-channel speech enhancement. The new method is a hybrid optimization algorithm, which employs the combination of the conventional θ-PSO and the shuffled sub-swarms particle optimization (SSPSO) technique. It is known that the θ-PSO algorithm has better optimization performance than standard PSO al...
متن کاملروشی نو برای حذف پژواک آکوستیکی استریو با استفاده از ساختار مبتنی بر الگوریتم وفقی با ورودی برش یافته
Stereophonic acoustic echo cancellation is one of the expanding areas in the field of the speech/multimedia communication systems. In a conventional stereophonic acoustic echo canceller, working in a stereo communication system, the existence of the strong cross-correlation between the input signals to the two channels has the problem of the low convergence speed of the weights of the adaptive ...
متن کاملSpeech Enhancement using Adaptive Data-Based Dictionary Learning
In this paper, a speech enhancement method based on sparse representation of data frames has been presented. Speech enhancement is one of the most applicable areas in different signal processing fields. The objective of a speech enhancement system is improvement of either intelligibility or quality of the speech signals. This process is carried out using the speech signal processing techniques ...
متن کاملUtilizing Kernel Adaptive Filters for Speech Enhancement within the ALE Framework
Performance of the linear models, widely used within the framework of adaptive line enhancement (ALE), deteriorates dramatically in the presence of non-Gaussian noises. On the other hand, adaptive implementation of nonlinear models, e.g. the Volterra filters, suffers from the severe problems of large number of parameters and slow convergence. Nonetheless, kernel methods are emerging solutions t...
متن کاملOn A New Hybrid Speech Coder using Variables LPF
To encode the speech quality with reduce the redundancy within samples that resulted from domain processing method like PCM and LPC, Source coding or Waveform coding methods can be considered. However, it is well known that when conventional sampling methods are applied directly to speech signal, the required amount of data is comparable to or more than that of uniform sampling method. To overc...
متن کامل