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On the duality between line-spectral frequencies and zero-crossings of signals

IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2001年第4期9卷 458-461页

作者： Kumaresan, R Wang, YD Univ Rhode Isl Dept Elect Engn Kingston RI 02881 USA

Line spectrum pairs (LSPs) are the roots (located in the complex-frequency or z-plane) of symmetric and antisymmetric polynomials synthesized using a linear prediction (LPC) polynomial. The angles of these roots, known as line-spectral frequencies (LSFs), implicitly represent the LPC polynomial and hence the spectral envelope of the underlying signal. BY exploiting the duality between the time and frequency domains, we define analogous polynomials in the complex-time variable zeta. The angles of the roots of these polynomials in zeta -plane now correspond to zero-crossing time instants. Analogous to the fact that the line-spectral frequencies represent the spectral envelope of a signal, these zero-crossing locations can be used to represent the temporal envelope of bandpass signals.

关键词： Polynomials linear predictive coding Signal synthesis Frequency synthesizers Speech synthesis Band pass filters Frequency domain analysis Speech coding Speech recognition Quantization

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Efficient code excited linear predictor using redundant vector quantiser representations

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ELECTRONICS LETTERS 2001年第22期37卷 1370-1372页

作者： Krishnan, V Mikhael, WB Univ Cent Florida Coll Engn & Comp Sci Dept Elect Engn Orlando FL 32816 USA

A novel linear prediction (LP) model-bused coding technique is presented. where the advantages of multiple non-orthogonal domain representations of LP coefficients and the residuals are exploited in conjunction with vector quantisation. The proposed technique is applied to speech signals and the resulting performance improvement is clearly demonstrated in terms of the reconstruction quality.

关键词： LP model-based coding technique speech codecs LP coefficients speech coding vector quantiser representations Signal processing theory residuals reconstruction quality error compensation efficient predictor redundant VQ representations linear predictive coding code excited linear predictor codec vector quantisation redundancy Speech and audio coding Signal processing and detection multiple nonorthogonal domain representations speech signals

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Multi stage matrix quantization for very low bit rate speech coding 3

Multi stage matrix quantization for very low bit rate speech...

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3rd IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC 01)

作者： Ozaydin, S Baykal, B Gazi Univ Dept Elect & Elect Engn Ankara Turkey

ISBN: (纸本)0780367200

Efficient quantization of linear predictive coding (LPC) filter coefficients play an essential role in very low bit rate speech coding systems. This paper examines a new suboptimal matrix quantization scheme for LPC parameters, called multi stage matrix quantization (MSMQ), which operates at bit rates between 400 and 800 bits/s, With the new matrix quantization method, using a 22,5 ms LPC analysis frame, spectral distortion about 1 dB is achieved at 800 bits/s, In the proposed coder, line spectral frequency (LSF) parameters of multiple consecutive frames are grouped into a superframe and jointly quantized. The new residual LSF vector quantization scheme gives a bit rate reduction in MSMQ without any additional complexity or storage. The new MSMQ leads into several schemes of various computational complexity/storage characteristics.

关键词： Bit rate Computational complexity Frequency Large scale integration linear predictive coding Nonlinear filters Spectral analysis Speech coding Vector quantization Wireless communication

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Speech recognition chip for monosyllables

Speech recognition chip for monosyllables

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Asia and South Pacific Design Automation Conference (ASP-DAC 2001)

作者： Nakamura, K Zhu, Q Maruoka, S Horiyama, T Kimura, S Watanabe, K Nara Inst Sci & Technol Grad Sch Informat Sci Nara 6300101 Japan

ISBN: (纸本)0780366336

In the paper, we present a real-time speech recognition chip for monosyllables such as A, B,.,., etc. The chip recognizes up to 64 monosyllables based on the Hidden Markov Model (HMM), which is a well known speaker-independent recognition method. The chip accepts a short-speech frame including 256 16-bit digitized samples corresponding to 11.6 msec period, and outputs the 6-bit symbol code of monosyllables for 16 short-frames (corresponding to 185.6 msec), A learning circuit to update HMM parameters for the recognition chip has also been designed, and the recognition chip includes an interface to the learning circuit. We have fabricated the recognition chip by VDEC Rohm 0.6 mum process on a 4.5 mm x 4.5 mm chip. We have also made a layout of the entire circuit including the learning circuit by VDEC Rohm 0.35 mum process on a 4.9 mm x 4.9 mm chip.

关键词： Algorithm design and analysis Circuits Feature extraction Hardware Hidden Markov models linear predictive coding Speech analysis Speech coding Speech recognition Vectors

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Candidate scheme for MP-MLQ search in G.723.1 3

Candidate scheme for MP-MLQ search in G.723.1

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3rd IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC 01)

作者： Chen, FK Yang, JF Natl Cheng Kung Univ Dept Elect Engn Tainan 70101 Taiwan

ISBN: (纸本)0780367200

For the purpose of low-cost or low power consumption, a cost-effective solution with good trade-off between quality and complexity must be estimated for small mobile machine. In this paper, we propose a condensed MP-MLQ algorithm to reduce search computation required for G723.1. Simulation results show that the proposed method can reduce about 50 similar to 82% computational load for stochastic codebook search with perceptually intangible degradation. This reductionism would be helpful to save the power consumption and relax CPU load for the mobile-machine in wireless communications, e.g. PALM, PDA.

关键词： Computational modeling Degradation Distributed computing Energy consumption GSM linear predictive coding Quantization Speech analysis Speech coding Stochastic processes

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A real-time 64-monosyllable recognition LSI with learning mechanism

A real-time 64-monosyllable recognition LSI with learning me...

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Asia and South Pacific Design Automation Conference (ASP-DAC 2001)

作者： Nakamura, K Zhu, Q Maruoka, S Horiyama, T Kimura, S Watanabe, K Nara Inst Sci & Technol Grad Sch Informat Sci Nara 6300101 Japan

ISBN: (纸本)0780366336

In the paper, a real-time 64-mono-syllable recognition LSI is presented. The LSI accepts 11.6 msec speech frame and outputs a 6-bit symbol-code for each frame by the end of the next frame with the pipelining manner. The recognition method is based on the Hidden Markov Model and is speaker-independent. An on-chip learning mechanism has also been designed, but the circuit is off-chip at present implementation because of the restriction of LSI area, The LSI is fablicated by VDEC Rohm with 0.6 mum process on a 4.5 mm x 4.5 mm chip.

关键词： Adders Circuits Hardware Hidden Markov models Large scale integration Learning systems linear predictive coding Speech analysis Speech coding Speech recognition

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Erasure-robust speech coding and concealment in VoIP systems with frame packetization

Erasure-robust speech coding and concealment in VoIP systems...

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IEEE Emerging Technologies Symposium on Broadband Communications for the Internet Era Symposium

作者： J. Wang J.D. Gibson Department of Electrical Engineering Southern Methodist University Dallas TX USA

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.

关键词： Speech coding Internet telephony Decoding linear predictive coding Degradation Bandwidth Robustness Interpolation Filters Books

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Adaptive underwater target classification with multi-aspect decision feedback

Adaptive underwater target classification with multi-aspect ...

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International Joint Conference on Neural Networks (IJCNN)

作者： M.R. Azimi-Sadjadi A.A. Jamshidi G.J. Dobeck Department of Electrical and Computer Engineering Colorado State University Fort Collins CO USA NSWC-Dahlgren Division Coastal Systems Station Signal and Image Processing Group Panama

ISBN: (纸本)0780370449

This paper presents a new scheme for underwater target classification in a changing environment. An adaptive target classification system is developed that uses the decisions of multiple aspects of the objects. The system employs a decision feedback mechanism to map the changed feature vector to a new feature space familiar to the classifier. Results on an acoustic backscattered data set, namely the 40 kHz data collected at Coastal Systems Station (CSS), are presented. This data set contains returns from six different objects at 72 aspect angles with 5 degrees separation and with varying signal-to-reverberation ratio (SRR). The results are then benchmarked with those of a neural network-based multi-aspect fusion system.

关键词： Feedback Adaptive systems Neural networks Sea measurements Cascading style sheets Reverberation Robustness linear predictive coding Testing Chirp modulation

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On extensive pump handling of chemical-mechanical polishing slurries

On extensive pump handling of chemical-mechanical polishing ...

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IEEE/SEMI Conference and Workshop on Advanced Semiconductor Manufacturing

作者： R.K. Singh B.R. Roberts Chemical Management Division BOC Edwards Santa Clara CA USA

ISBN: (纸本)0780365550

The effects of extensive pump handling on chemical-mechanical polishing (CMP) slurries have been investigated in this study. A number of common CMP slurries were recirculated in bellows pump loops. Large particle counts (LPC) and particle size distributions (PSDs) were generated using commercially available instruments. The PSDs of oxide CMP slurries illustrate distinct, developing characteristics as a result of repeated shearing during handling. Unlike most silica oxide slurries, alumina- and ceria-based slurries do not seem to generate large particles due to pump handling over extended periods of time. Some of these slurries show a decrease in number of large particles or soft flakes in pump recirculation tests. The PSD and LPC data for an oxide slurry handling test in a vacuum-pressure-dispense-technology-based pump show generation of fewer large particles as compared to a bellows pump. This study also demonstrates that insignificant progressive settling of abrasive particles takes place inside the pump or in the pump loop during handling of quick-settling CMP slurries.

关键词： Chemicals Slurries linear predictive coding Bellows Shearing Silicon compounds Testing Inorganic materials Solids Vacuum systems

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Single and double frame quantization of LSF parameters using noise feedback coding

Single and double frame quantization of LSF parameters using...

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IEEE International Conference on Communications (ICC)

作者： A.R. Fazel A.K. Khandani Department of Elec. and Comp. Engineering University of Waterloo Waterloo ONT Canada

We present a scheme for single frame (20 msec) and double frame (40 msec) quantization of line spectral frequency (LSF) parameters in a code-excited linear prediction (CELP) speech coder using noise feedback coding. To improve the performance, an appropriate lattice structure is used as the quantizer in the noise feedback loop. We also consider a switched structure based on using either a double frame quantizer or two single frame quantizers for two subsequent frames where an extra bit is used to specify the choice offering a lower distortion. Numerical results are presented showing an excellent performance with very low complexity.

关键词： Quantization Distortion measurement Delay Filters linear predictive coding Speech coding Speech analysis Euclidean distance Speech enhancement Feeds

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