Line spectrum frequencies (LSF) have been the prevailing parameter set to represent LPC coefficients in speech coding. Extensive research has been performed to exploit their interframe and intraframe correlations and ...
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Line spectrum frequencies (LSF) have been the prevailing parameter set to represent LPC coefficients in speech coding. Extensive research has been performed to exploit their interframe and intraframe correlations and quantize them more efficiently. Interframe coding of the LSF can cause error propagation when frame erasures occur. Since most LSF quantizers were designed with the primary concerns of bit-rate and complexity, less attention was paid to error propagation. We investigate the erasure performance of interframe LSF coding and compare it with an intraframe coding method. Our results show that with only 5% extra bit-rate, intraframe coding is much more robust to frame erasures and a typical improvement of 0.5 dB on spectral distortion can be obtained with 20% packet loss. Subjective listening tests indicate significant improvement as well.
We consider enhancement of speech based on a harmonic plus noise representation of the signal. We propose a novel model and its associated estimation procedure for the harmonic component. The model is nonparametric an...
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We consider enhancement of speech based on a harmonic plus noise representation of the signal. We propose a novel model and its associated estimation procedure for the harmonic component. The model is nonparametric and capable of representing nonstationarity and the estimation approach uses a data-driven method to optimize the duration from which each harmonic component is locally estimated.
We propose a new digital modulation classification method based on the continuous-time wavelet transformation (CWT) and the linear predictive coding (LPC) method. The LPC coefficients extracted from the LPC model of t...
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We propose a new digital modulation classification method based on the continuous-time wavelet transformation (CWT) and the linear predictive coding (LPC) method. The LPC coefficients extracted from the LPC model of the CWT for a modulated signal is chosen as the feature used to classify the modulation types of BPSK, QPSK, FSK and jammer. By using several reference features per modulation type we can make our algorithm robust to the influence of noise. To verify the proposed modulation classification algorithm, simulations are performed, which demonstrate excellent classification rates.
The output conductance of double-gate SOI transistors (DG-MOSFETs) exhibits a dependence versus frequency. This behavior is shown to be related to the loss mechanism due to capture and emission of carriers by interfac...
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The output conductance of double-gate SOI transistors (DG-MOSFETs) exhibits a dependence versus frequency. This behavior is shown to be related to the loss mechanism due to capture and emission of carriers by interface traps. The interface trap behavior can be represented by an equivalent capacitance C/sub it/ and resistance R/sub it/ model. We have shown that the EKV model combined with SILVACO simulations fully accounts for (i) the pinch-off voltage variation with drain voltage, (ii) the effect of the interface trap density on the output conductance (Vandooren et al., 1999), and (iii) the conductance degradation with total radiation dose. This paper investigates the frequency behavior of the output conductance observed in double-gate MOSFETs. We demonstrate that this behavior can be reproduced by including the frequency properties of interface traps in the EKV model.
Diversity schemes include information about packet n in future packets or send information about packet n via separate paths. If packet n is lost, it is reconstructed from information included in future packets or inf...
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Diversity schemes include information about packet n in future packets or send information about packet n via separate paths. If packet n is lost, it is reconstructed from information included in future packets or information received via separate paths. This paper presents CELP-based diversity schemes for voice over packet applications. The diversity schemes reduce the impact of packet losses while being efficient in terms of both bandwidth requirement and computational complexity. With our diversity schemes, transmission schemes that allocate bandwidth resources among diversity stages during congestion give significantly better performance than schemes that use no diversity during congestion, for the same bandwidth usage.
This paper uses a method of incorporating simultaneous masking into the calculation of a linearpredictive filter (SMLPC) as the front end to a 2 kbps waveform interpolation (WI) speech coder. A modification to the ma...
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This paper uses a method of incorporating simultaneous masking into the calculation of a linearpredictive filter (SMLPC) as the front end to a 2 kbps waveform interpolation (WI) speech coder. A modification to the masking threshold calculation used in SMLPC is proposed. This modification improves the performance of SMLPC in noise like sections by placing greater emphasis on strongly voiced speech. MOS test results reveal that the modified SMLPC improved the perceptual quality of the WI coder. The improvement is significant for female speakers whilst the quality for male speech is virtually unchanged. This result conflicts with previous results reported for SMLPC where only male speech was improved. The change is attributed to the modification of the masking threshold and confirms that adapting the masking threshold according to the pitch of the speech will allow SMLPC to remove more perceptually important information from all input speech than standard LPC.
In this paper, an information theoretic study of properties of the speech spectrum process is performed. Various techniques to model the probability density function are applied to the spectrum source to compute rate-...
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In this paper, an information theoretic study of properties of the speech spectrum process is performed. Various techniques to model the probability density function are applied to the spectrum source to compute rate-distortion functions. We estimate the difference in the required rate to achieve a given distortion for three different scenarios: interframe gain exploitation, low-pass filtering of LPC vectors and increased speech signal bandwidth. We obtain fairly consistent results for the different methods of calculating rate-distortion functions. The results show that for close to transparent LPC quantization we gain 4-6 bits per frame by exploiting first order interframe correlation. The new idea of using low-pass filtered LPC vectors has shown to decrease the coding cost with 1-3 bits per frame, depending on the cutoff frequency.
An approach for reducing the complexity of the switched-adaptive interframe vector prediction (SIVP) that is used for coding speech spectrum envelopes is proposed in this paper. To facilitate the search through the se...
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An approach for reducing the complexity of the switched-adaptive interframe vector prediction (SIVP) that is used for coding speech spectrum envelopes is proposed in this paper. To facilitate the search through the set of switched predictors used for prediction of the input LSF (line spectral frequency) vector, the predictors are organized in a binary tree structure. For a conventional full-searched SIVP coder with N=2/sup b/ predictors, predictions must be performed by all of them in order to determine the best one, while only 2b predictions are sufficient for the proposed binary tree-searched coder. A design procedure for obtaining optimal binary tree-structured predictors is given. The effectiveness of the proposed coder is evaluated and the results are compared to the baseline full-searched coders as a function of the number of predictors and the resolution of the vector quantizers used for quantization of the prediction residual. A discussion of possible applications of the proposed coder is also given.
An overview of the requirements of digital covert audio acquisition (DCAA) is provided and a scalable hybrid coder tuned for the coding of intelligible speech in a covert environment is proposed. The codec incorporate...
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An overview of the requirements of digital covert audio acquisition (DCAA) is provided and a scalable hybrid coder tuned for the coding of intelligible speech in a covert environment is proposed. The codec incorporates linear predictive coding (LPC) and an M-band discrete wavelet transform (DWT) to offer effective intelligible speech coding in the presence of multiple signal sources at bit rates between 8 kbps and 32 kbps. The results of informal intelligibility testing and an analysis of the algorithm's complexity are presented to demonstrate the performance of the proposed coder.
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