This paper describes a new technique, called the empirical mode decomposition (EMD) that has recently been pioneered by N. E. Huang and al., for adaptively representing nonstationary signals as sums of zero-mean AM-FM...
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This paper describes a new technique, called the empirical mode decomposition (EMD) that has recently been pioneered by N. E. Huang and al., for adaptively representing nonstationary signals as sums of zero-mean AM-FM components [N. E. Huang, et al., 1998]. The components, called intrinsic mode functions (IMFs), allow the analysis of frequency composition of one-dimensional signals. Applied to speech signal, the EMD allows us to study the different intrinsic oscillatory modes. Besides, computing the LPC analysis of each mode provides an estimation of formants. The presented method is firstly applied on a sum of pure frequency signals. Among different modes we can detect all frequencies taking a part of a signal.
Fault diagnosis and monitoring of the machines operation in the power plants play an important role in safety operation and maintenance of those operating machines. In this paper we propose the fault diagnosis algorit...
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Fault diagnosis and monitoring of the machines operation in the power plants play an important role in safety operation and maintenance of those operating machines. In this paper we propose the fault diagnosis algorithm using the LPC coefficients with sound acquisition system from the operating machines through the single LPC spectrum is possible.
Results obtained in an experimental evaluation of speech intelligibility of an adaptive processing technique designed to enhance the intelligibility of speech in noise are presented. The technique relies on speech fea...
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Results obtained in an experimental evaluation of speech intelligibility of an adaptive processing technique designed to enhance the intelligibility of speech in noise are presented. The technique relies on speech features extracted from noise-corrupted speech to control a cepstral processor. The speech-processing algorithm has been improved over the fundamental technique described by R.J. Conway and R.J. Niederjohn (1987). The evaluation method makes use of a computer implementation of a modified version of the diagnostic rhyme test. Results for four subject and several signal-to-noise ratios are presented.< >
A fast adaptive method for tracking the roots of a time-varying complex domain polynomial is derived. The approach uses the method of homotopy continuation and is efficient from both mathematical and implementation st...
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A fast adaptive method for tracking the roots of a time-varying complex domain polynomial is derived. The approach uses the method of homotopy continuation and is efficient from both mathematical and implementation standpoints. The method is globally convergent and tracks all roots simultaneously. An example that verifies the accurate tracking ability of the algorithm is presented. Applications which could benefit from this method are also discussed.< >
This paper presents a 1.2 kbps speech coder based on the mixed excitation linear prediction (MELP) analysis algorithm. In the proposed coder, the MELP parameters of three consecutive frames are grouped into a superfra...
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This paper presents a 1.2 kbps speech coder based on the mixed excitation linear prediction (MELP) analysis algorithm. In the proposed coder, the MELP parameters of three consecutive frames are grouped into a superframe and jointly quantized to obtain a high coding efficiency. The interframe redundancy is exploited with distinct quantization schemes for different unvoiced/voiced (U/V) frame combinations in the superframe. Novel techniques for improving performance make use of the superframe structure. These include pitch vector quantization using pitch differentials, joint quantization of pitch and U/V decisions and LSF quantization with a forward-backward interpolation method. Subjective test results indicate that the 1.2 kbps speech coder achieves approximately the same quality as the proposed federal standard 2.4 kbps MELP coder.
To investigate the neural efficiency theory of intelligence, electroencephalograms (EEG) were recorded while 15 intellectually gifted children and 15 average children performed a 2-back working memory task. The amplit...
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ISBN:
(纸本)9781424445424
To investigate the neural efficiency theory of intelligence, electroencephalograms (EEG) were recorded while 15 intellectually gifted children and 15 average children performed a 2-back working memory task. The amplitude of P2, N2, and LPC were analyzed. The results showed that intellectually gifted children performed more accurately and had larger LPC mean amplitudes than their intellectually average peers under the matching condition, suggesting that intellectually gifted individual can use their brain and allocate cognitive resources more efficiently.
The transmission of compressed audio and video data over ATM packet networks is quickly becoming a practical way to implement a multimedia teleconferencing system. To reduce the required bit rate of the combined audio...
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The transmission of compressed audio and video data over ATM packet networks is quickly becoming a practical way to implement a multimedia teleconferencing system. To reduce the required bit rate of the combined audio/video sequence to that of basic rate ISDN, we encode the video using a subband coder and encode the audio using a CELP coder. In this paper, we examine the effects that packet losses, and the variable delay and delay jitter of ATM networks have on our audio coding system. Results for a large time-bandwidth product network are presented. Specifically, ways to make the speech coding more robust to packet loss are examined.
In order to deliver real time, high quality voice services in packet based voice system (e.g. voice over Internet protocol, VoIP) system designers must tackle inherent quality problems related to possible packet loss....
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In order to deliver real time, high quality voice services in packet based voice system (e.g. voice over Internet protocol, VoIP) system designers must tackle inherent quality problems related to possible packet loss. To combat the inevitable speech quality deterioration resulting from the loss of transmitted packets of speech information, techniques that provide estimates of the lost information that is needed by the speech recovery process are of considerable interest. Furthermore, in VoIP systems employing linear predictive coding (LPC) based speech coders, a significant percentage of the coded speech information represent the values of LPC coefficients and thus a new approach for estimating missing LPC filter coefficients is presented in this paper. This approach employs a new formulation of LSP recovery system architecture where evolving fuzzy rule-based models and particularly so-called evolving Takagi-Sugeno models are deployed to generate the required estimates of missing LSPs. The proposed missing parameters estimation technique is generic and initial experimental results demonstrate its considerable potential in improving the quality of LPC based decoded speech in VoIP applications
In voice over IP (VoIP) networks, multiple voice frames can be sent within one packet to increase the network efficiency. When packet loss happens, the voice decoder often tries to conceal the erased frames from recei...
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In voice over IP (VoIP) networks, multiple voice frames can be sent within one packet to increase the network efficiency. When packet loss happens, the voice decoder often tries to conceal the erased frames from received parameters. This paper discuss the quantitative dependency of concealment quality on the packetization rate in terms of LPC distortion. The performance of an erasure-robust. concealment method is analyzed and simulated in comparison with the standard G.729 concealment method. Results show that around 1.2 dB improvement on spectral distortion can be obtained with erasure robust. concealment compared with G.729 under 1-6 frames/packet, packetization. A method to determine the maximum packet size given the network load and expected concealment quality is introduced. The proposed method serves as the packetization guidance during the call setup procedure in VoIP sessions.
It is important to obtain effective feature values of data stream and forecast them in overload system for mining data stream, because data streams are often bursty and data characteristic vary over time. In this pape...
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It is important to obtain effective feature values of data stream and forecast them in overload system for mining data stream, because data streams are often bursty and data characteristic vary over time. In this paper, we introduce linear predictive coding (LPC) technology to obtain feature values using fewer coefficients. Generalized autoregressive conditional heteroscedastic (GARCH) -generalized regression neural network (GARCH-GRNN) model is used to forecast the feature values of which the data streams are shed, and we perform similarity search using these forecasting values. A load shedding framework based on LPC and GARCH-GRNN (LS-LG) for similarity search on data stream is constructed to achieve minimized mining loss. Experimental results indicate that LS-LG is an effective method in improving query quality when the system is under overload situation.
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