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IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： O. Lev D. Malah Dept. of Electr. Eng. Technion-Israel Inst. of Technol. Haifa Israel Department of Electrical Engineering Technion Israel Institute of Technology Israel

We present a low bit rate speech coder based on a long-term model (LTM) for voiced speech, and on the WI coder. In the LTM, a periodic input signal undergoes a time-varying spectral shaping representing the evolution of the pitch-cycle waveform. The resulting signal, which has a fixed pitch period but a time-varying pitch-cycle waveform, is multiplied by a time-varying gain function that represents the variation in speech loudness. The resulting signal then undergoes a time-axis warping, which represents the evolution of the pitch period, yielding the output speech signal. The spectral shaping in the proposed coder is based on WI. In WI, speech (or LPC residual) is observed as a continuously evolving sequence of pitch cycle waveforms. A subset of these waveforms is extracted and coded. In the decoder, after inverse quantization, missing waveforms are synthesized by interpolation. The extracted waveforms are normalized to a fixed length and sequentially aligned using a cyclical shift. Then, a two-dimensional surface, called prototype waveform surface or characteristic waveform (CW) is produced from these waveforms.

关键词： Speech analysis Speech synthesis Surface waves Interpolation linear predictive coding Decoding Energy states Gaussian noise Signal analysis Bit rate

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Voice morphing by 3-D waveform interpolation surface and lossless tube area function

Voice morphing by 3-D waveform interpolation surface and los...

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IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： G. Porat Y. Lavner Signal and Image Processing Laboratory (SIPL) Faculty of Electrical Engineering Technion-Israel Institute of Technology Haifa Israel Computer Science Tel-Hai Academic College Upper Galilee Israel

Voice morphing is the process of gradually transforming the voice of a given speaker to that of another. The ability to change the speaker's individual characteristics and produce high-quality voices can be used in many applications. For example, in multimedia and video entertainment, voice morphing is just like its visual counterpart: while seeing a face gradually changing from one person's to another's, we can simultaneously hear the voice changing as well. Another application could be in forensic voice identification: creating a voice-bank of different pitches, rates, and timbres, to assist in recognition of the suspect's voice. In this study we present a new technique, which enables the production of N intermediate voices that gradually change between voices of two speakers, or one voice signal that changes gradually. This technique is based on two components. One is creating a 3D prototype waveform interpolation (PWI) surface from the residual error ' signal, which is estimated from LPC analysis, to produce a new intermediate excitation signal. The second component is a representation of the vocal tract by a lossless tube area function, and interpolation of the two speakers' parameters.

关键词： Interpolation Surface waves Speech analysis Signal analysis Surface reconstruction Prototypes Filters linear predictive coding Speech coding Signal synthesis

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Joint MELP turbo code with unequal error protection

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ELECTRONICS LETTERS 2001年第10期37卷 637-639页

作者： Wang, Q Koh, SN Nanyang Technol Univ Sch Elect & Elect Engn Singapore 639798 Singapore

The mixed excitation linear prediction (MELP) algorithm has been recently selected as the new federal standard for 2.4kbit/s coding of speech signals. The authors exploit the average residual inter-frame correlation and the error sensitivities of the bits in a MELP frame to enhance the robustness of the proposed joint MELP turbo coding schemes for Operations over Rayleigh fading channels.

关键词： BER speech coding mixed excitation linear prediction algorithm error statistics error detection combined source-channel coding turbo codes Codes unequal error protection correlation theory linear predictive coding residual inter-frame correlation Rayleigh channels bit error sensitivities Speech and audio coding Rayleigh fading channels Information theory joint MELP turbo code 2.4 kbit/s error correction MELP algorithm speech signals

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

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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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