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检索条件"主题词=Linear Predictive coding"

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Efficient parameter quantisation for 2.4/1.2 kb/s split-band LPC coding

Efficient parameter quantisation for 2.4/1.2 kb/s split-band...

引用

IEEE Workshop on Speech coding Proceedings

作者： S. Villette Y.D. Cho A.M. Kondoz Centre for Communication Systems Research University of Surrey Guildford Surrey UK

Speech coding at very low bit rates has many applications such as answering machines, IP telephony, mobile communications, military communications etc. Most low bit rate coders operate at around 2.4 kb/s, as the speech quality degrades too much below this bit rate. We describe a frequency domain speech coder capable of operating at both 2.9 and 1.2 kb/s, and produces good quality synthesised speech. Both rates use the same analysis and synthesis building blocks over 20 ms, but the 1.2 kb/s coder jointly quantises three sets of parameters every 60 ms to reduce the bit rate while maintaining speech quality. We also describe the quantisation methods used to lower the bit rate from 2.4 kb/s to 1.2 kb/s while retaining most of the quality of the higher bit rate version.

关键词： Quantization linear predictive coding Bit rate Speech synthesis Speech analysis Speech coding Telephony Mobile communication Military communication Degradation

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GSM speech coding and speaker recognition

GSM speech coding and speaker recognition

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： L. Besacier S. Grassi A. Dufaux M. Ansorge F. Pellandini CLIPSAMAG GEOD team Joseph Fourier University Grenoble France Institute of Microtechnology University of Neuchâtel Neuchatel Switzerland

This paper investigates the influence of GSM speech coding on text independent speaker recognition performance. The three existing GSM speech coder standards were considered. The whole TIMIT database was passed through these coders, obtaining three transcoded databases. In a first experiment, it was found that the use of GSM coding degrades significantly the identification and verification performance (performance in correspondence with the perceptual speech quality of each coder). In a second experiment, the features for the speaker recognition system were calculated directly from the information available in the encoded bit stream. It was found that a low LPC order in GSM coding is responsible for most performance degradations. By extracting the features directly from the encoded bit-stream, we also managed to obtain a speaker recognition system equivalent in performance to the original one which decodes and reanalyzes speech before performing recognition.

关键词： GSM Speech coding Speaker recognition Spatial databases Degradation Code standards linear predictive coding Data mining Feature extraction Decoding

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Single chip implementation of the 1.6 kbps speech vocoder

Single chip implementation of the 1.6 kbps speech vocoder

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Jia-Ching Wang Jhing-Fa Wang Han-Chiang Chen Department of Electrical Engineering National Cheng Kung University Tainan Taiwan

ISBN: (纸本)0780354826

In this paper, we propose a low bit rate speech vocoder and its corresponding VLSI implementation. The vocoder exploits the interpolation property so that the fine quality in synthesized speech is obtained even though the bit rate is as low as 1.6 kbps. Two novel methods including pitch detection and LSP decoding which are suitable for VLSI implementation are also proposed. The heuristic pitch detection algorithm avoids the heavy computational load introduced by the traditional normalized autocorrelation method. The memory storing triangular function value is no longer needed after adopting the new LSP decoding process. The chip is designed with area effective feature and is suitable for stand alone application.

关键词： Vocoders Bit rate Decoding Speech coding linear predictive coding Filters Very large scale integration Autocorrelation Interpolation Speech synthesis

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A cochlear prosthesis stimulation algorithm targeting to minimize the rehabilitation phase and enhance speech comprehension

A cochlear prosthesis stimulation algorithm targeting to min...

引用

Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： K. Gadhoumi J. Mouine Z. Chtourou Department of Electrical Engineering and Computer Science University of Sherbrooke Sherbrooke QUE Canada

ISBN: (纸本)0780364651

Efficiency of cochlear stimulation is closely related to the signal processing of the input speech. A variety of techniques were proposed and used for cochlear prosthesis speech analysers. So different seems to be, these techniques are based on two major approaches. The first one consists of stimulating the cochlea according to extracted speech features and the second is based on wide band speech characteristics. Clinically, it is well known that in their initial rehabilitation phase, new cochlear prostheses users face some problems of distinguishing what they hear. This is due to the infinity of stimulation set related to the infinite possibilities of the input speech. In order to overcome this problem and minimise the rehabilitation period, a new stimulation algorithm based upon vector quantization is proposed.

关键词： Prosthetics Speech processing Vector quantization Dictionaries predictive models linear predictive coding Speech analysis Distortion measurement Autocorrelation Signal processing algorithms

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Novel speech spectra all-pole modelling based upon selective even-samples linear prediction

Novel speech spectra all-pole modelling based upon selective...

引用

IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： K.F. Chang P. Cheong S.W. Ting K.W. Tam Faculty of Science and Technology University of Macau Macao China

ISBN: (纸本)0780365429

In this paper, a novel linear predictive method (SELP) for speech spectra modelling is proposed. This method allows one to develop an all-pole filter which combines p+1 consecutive even preceding samples of the speech signal x(n) into p pairs for linear extrapolation. In addition, a weighting selective scheme is employed to obtain high signal-to-error ratio (SER). Comparing to the traditional LPC modelling, the proposed filter's order is then raised to 2p+2 when both filters are with p-normal equations. In order to demonstrate the proposed method usefulness, this new model is simulated at 22.05 kHz speech spectra. Experimental results show that at least 10 dB SER improvement is obtained with p=5 when compared with that of LPC modelling.

关键词： predictive models linear predictive coding Extrapolation Frequency conversion Nonlinear filters Equations Speech recognition Spectral analysis Information rates Bit rate

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Shaped fixed codebook search for CELP coding at low bit rates

Shaped fixed codebook search for CELP coding at low bit rate...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： E. Erzin Lucent Technologies Inc. Whippany NJ USA

This paper presents a new shaped fixed codebook (FCB) search technique for code excited linear predictive (CELP) coding. The state of art CELP coding techniques operate at rates above 4.0 kbps, as it gets harder to build a good FCB contribution with a minimal bit budget. In this paper the shaped FCB search is presented to ease this problem, and achieve a better FCB contribution with a reduced bit budget. The shaped FCB search integrated to a 4 kbps CELP coder is presented and the subjective performance results are reported which show that the coder is significantly better than the IS-127 half rate coder at 4 kbps.

关键词： Bit rate Speech coding Books Pulse shaping methods Art Speech enhancement linear predictive coding Filters Speech codecs Code standards

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Quantization of line spectral parameters using a trellis structure

Quantization of line spectral parameters using a trellis str...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： F. Lahouti A.K. Khandani Department of E&CE University of Waterloo Waterloo ONT Canada

In this article, a low bit-rate low-complexity block-based trellis quantization (BTQ) scheme is proposed for quantization of line spectral frequencies for speech coding applications. The branches in the trellis diagram correspond to the LSF difference code-words and the states correspond to quantized LSF parameters. An efficient algorithm for the index generation (finding the index of a path in the trellis) is introduced. The proposed BTQ achieves the transparent coding quality at 23 bits/frame (1150 b/sec.) and offers a gain of 3 b/frame (150 b/s) and significant reduction in complexity, compared to the IS-641 split-VQ. An interframe BTQ scheme is also presented to exploit the redundancies between the adjacent frames. The interframe scheme is found to achieve an additional 50 b/sec reduction of the bit-rate and is based on adaptive block-based trellis quantization of the prediction residues.

关键词： Quantization Speech coding linear predictive coding Speech analysis Frequency response Nonlinear filters Signal analysis Random access memory Computational complexity Information technology

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Hybrid audio coding for speech and audio below medium bit rate

Hybrid audio coding for speech and audio below medium bit ra...

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IEEE International Conference on Consumer Electronics (ICCE)

作者： K. Makino J. Matsumoto Dev. Center Sony Corp. Tokyo Japan HomeNet Processing Laboratory Development CenterHome Network Company Sony Corporation USA

A novel a audio/speech coding algorithm, hybrid audio coding (HAC) is described. New features of the algorithm include window switching with generalized MDCT, an improved quantization scheme of the MDCT coefficients, and waveform normalization in the time domain. HAC provides a good quality at a bit rate of 8 to 16 kbps, and it is also proven that the developed algorithm is effective for both audio and speech signals.

关键词： Audio coding Speech coding Bit rate Codecs Testing linear predictive coding Quantization Multiple signal classification Decoding Smoothing methods

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Dependence of Channel Width and Length on MOSFET Matching for 0.18 um CMOS Technology

Dependence of Channel Width and Length on MOSFET Matching fo...

引用

European Conference on Solid-State Device Research (ESSDERC)

作者： R. Difrenza P. Llinares G. Ghibaudo E. Robillart E. Granger LPCS ENSERG Grenoble France STMicroelectronics Central Research and Development Laboratory Crolles France

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Distributed speech recognition services (DSRS)

Distributed speech recognition services (DSRS)

引用

International Symposium on Multimedia Software Engineering

作者： M. Coyner Tsang-Ming Jiang V. Volobouev Department of Mathematical Sciences University of Alaska Fairbanks AK USA

Distributed speech recognition services (DSRSs) provide an anytime, anywhere and any-device speech recognition environment that is intelligent enough to interact with users in a more natural manner. The primary goal is to provide users with the ability to dictate commands and/or documents among other potential services. The system coordinates the efforts of applications running in a distributed environment. For example, a user is able to dictate a document using their local word processor and a DSRS's remotely located speech engine. DSRSs encourage cooperation among individual programs in order to combine the efforts of individual applications to fulfill a user's request.

关键词： Speech recognition Search engines Hidden Markov models Application software Speech processing Wireless application protocol Personal digital assistants World Wide Web linear predictive coding Cepstral analysis

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