This paper presents a set of low-complexity tools used in lossless coding of G.711 bitstream, based on linear prediction. One is an algorithm for quantizing the PARCOR/reflection coefficients and the other is an estim...
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This paper presents a set of low-complexity tools used in lossless coding of G.711 bitstream, based on linear prediction. One is an algorithm for quantizing the PARCOR/reflection coefficients and the other is an estimation method for the optimal prediction order. Both tools are based on a criterion that minimizes the entropy of the prediction residual signal and can be implemented in fixed-point arithmetic at very low-complexity. Since proposed methods show efficient performance in terms of compression and complexity, they are adopted in the Recommendation ITU-T G.711.0, a new standard for lossless compression of G.711 (A-law/mu-law logarithmic PCM) payload.
This paper presented the modification to Esophageal Speech (ES) enhancement using Adaptive Gain Equalizer (AGE) for modifying the voicing source. However, the voicing source used previously with AGE, obtained using co...
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ISBN:
(纸本)9781479947966
This paper presented the modification to Esophageal Speech (ES) enhancement using Adaptive Gain Equalizer (AGE) for modifying the voicing source. However, the voicing source used previously with AGE, obtained using conventional linear Predication (LP) vocal tract transfer function (AGE-LP), has produced low quality speech due to sensitivity to background noise. The better quality ES can be obtained by estimating voicing source through Iterative Adaptive Inverse Filtering utilized Weighted linear Prediction (WLP) vocal tract transfer function (AGE-IAIF). The system performance evaluated through Harmonic to Noise Ratio (HNR), and system has shown 3 dB enhancement by AGE-IAIF over previously enhancement method AGE-LP.
The aim of this work is the design of a novel audio watermarking technique based on the linear predictive coding approach and on the psychoacoustic model defined in the MPEG-I standard. The embedding imperceptibility ...
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ISBN:
(纸本)9781467361415
The aim of this work is the design of a novel audio watermarking technique based on the linear predictive coding approach and on the psychoacoustic model defined in the MPEG-I standard. The embedding imperceptibility is achieved by exploiting the frequency masking effect. The evaluation of the perceived audio quality is performed by means of the standardized ITU-R BS.1387 quality assessment method. The effectiveness of the proposed method is tested in presence of attacks.
We explicitly construct binary measurement matrices with good sparse approximation guarantees. Specifically, our measurement matrices have an order optimal number of measurements and have l(1)/l(1) approximation guara...
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ISBN:
(纸本)9781479913213
We explicitly construct binary measurement matrices with good sparse approximation guarantees. Specifically, our measurement matrices have an order optimal number of measurements and have l(1)/l(1) approximation guarantee. Our construction uses the progressive edge growth technique. We apply coding theoretic results and rely on a recent connection of compressed sensing to LP relaxation for channel decoding.
The objective of this paper is to provide feature extraction algorithm for underwater targets. The targets are homogeneous elastic bodies of finite dimensions. The targets considered are a brass sphere, a PVC sphere, ...
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ISBN:
(纸本)9789380095455
The objective of this paper is to provide feature extraction algorithm for underwater targets. The targets are homogeneous elastic bodies of finite dimensions. The targets considered are a brass sphere, a PVC sphere, a brass cylinder, a PVC cylinder, concrete block and MS cylinder of different dimensions. The incident acoustic signal used was a linear frequency modulated (LFM) signal of finite duration with the signal bandwidth of 40 kHz to 80 kHz. The scattered acoustic signal from the targets are recorded and processed for feature selection. The scattered signals were analysed using power spectrum analysis, linear predictive coding and Auto Regressive (AR) modelling, and its statistical features are extracted for all the targets. The nature of the backscattered signal for the underwater targets is also explained. The extracted features are passed into the feed forward neural network (FFNN) classifier. FFNN was used to identify the targets of six classes, to check the validity of extracting the feature of the targets. The result of the neural network shows that this feature extraction algorithm could enhance the fractal features of the signals and reduce the number of dimensions of the feature space and prove that it can efficiently classify underwater targets. A comprehensive study was then carried out to compare the classification performance by using these data sets in terms of performance analysis like specificity and sensitivity.
A low delay audio coding scheme with good perceptual audio quality for a desired limited bit rate is presented. The proposed audio coding scheme is based on differential pulse code modulation (DPCM) and block compande...
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ISBN:
(纸本)9781479908332
A low delay audio coding scheme with good perceptual audio quality for a desired limited bit rate is presented. The proposed audio coding scheme is based on differential pulse code modulation (DPCM) and block companded (BC) quantization. Prediction is realized as a FIR filter in lattice structure. DPCM performs in feedback manner, therefore no transmission of prediction filter coefficients is needed. The incorporation of BC quantization in the DPCM relies on a prediction error recalculation scheme. The use of BC quantization in the DPCM allows to accurately follow the prediction error signal. This improves the perceptual audio quality significantly compared to a plain DPCM with an adaptive quantizer. An algorithmic delay below a half millisecond and an overhead of less than a half bit per sample is introduced due to the short fixed block length of the BC quantizer. Therefore, a real time bidirectional audio application is achievable.
The navigation autopilot system was developed using real-time voice command recognition system for the ship open water cruise. In this design, linear predictive coding (LPC) algorithm is used for the extraction of the...
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ISBN:
(纸本)9781467355636;9781467355629
The navigation autopilot system was developed using real-time voice command recognition system for the ship open water cruise. In this design, linear predictive coding (LPC) algorithm is used for the extraction of the voice command feature and Dynamic Time Warping (DTW) algorithm is used for attribute matching on MATLAB program.
This paper presents the investigation of acoustic characteristics of ethnically diverse accents in Malaysian English across genders. The study of Malaysian English is still at infancy and the cues of how accents can b...
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This paper presents the investigation of acoustic characteristics of ethnically diverse accents in Malaysian English across genders. The study of Malaysian English is still at infancy and the cues of how accents can be differentiated by human are less understandable. Understanding this through the use of formants would discover the importance of these features that can be used to drive the classification results. It was found that males and females differ in terms of all formants scores that correlate to accents in great details using two-way and one-way analysis of variance and the plots of normal fit of individual formant.
In Mixed Excitation linear Prediction algorithm (MELP), the sub-band Unvoiced/Voiced parameters play an important role in improving the naturalness of synthetic speech. However, the coding efficiency with five bits pe...
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ISBN:
(纸本)9781479905607
In Mixed Excitation linear Prediction algorithm (MELP), the sub-band Unvoiced/Voiced parameters play an important role in improving the naturalness of synthetic speech. However, the coding efficiency with five bits per frame brings difficulties for very low bit rate speech coding. In this paper, the three consecutive MELP frames are grouped into a super-frame, and the fifteen dim sub-band Unvoiced/Voiced parameters are quantized. Through counting the Unvoiced/Voiced distribution probability and optimizing the codebook designed by the distortion measure, it is implemented that every fifteen dim Unvoiced/Voiced vector is quantized efficiently with three bits for each super-frame. Simulation results show that the intelligibility and naturalness are efficiently maintained for synthesis speech, and the quantization scheme can be widely applied to speech coding algorithm below 600bps.
The navigation autopilot system was developed using real-time voice command recognition system for the ship open water cruise. In this design, linear predictive coding (LPC) algorithm is used for the extraction of the...
详细信息
The navigation autopilot system was developed using real-time voice command recognition system for the ship open water cruise. In this design, linear predictive coding (LPC) algorithm is used for the extraction of the voice command feature and Dynamic Time Warping (DTW) algorithm is used for attribute matching on MATLAB program.
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