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检索条件"主题词=Linear predictive coding"

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Analysis by synthesis speech coding with generalized pitch prediction

Analysis by synthesis speech coding with generalized pitch p...

引用

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

作者： P. Mermelstein Y. Qian INRS-Télécommunications Verdun QUE Canada

A new analysis-by-synthesis speech coding structure is presented for high-quality speech coding in the 4 to 8 kb/s range. CELP with generalized pitch prediction (GPP-CELP) differs from classical code-excited linear prediction (CELP) in that for voiced segments it is the speech signal that is decomposed into a component predictable with the aid of the adaptive codebook (ACB) and a nonpredictable aperiodic component, not the LPC residual. The spectrum of the aperiodic component is estimated by linear-prediction analysis. An approximation to the aperiodic component is synthesized from a stochastic codebook of sparse pulse sequences and its spectrum is shaped by the LPC synthesis filter. The ACB contains samples of the past reconstructed signal, low-passed to increase the pitch prediction gain. For voiced segments the new structure yields higher pitch prediction gain and lower linear-prediction gain than classical CELP. Subjective and objective comparisons reveal significant advantages for GPP-CELP over classical CELP.

关键词： Speech analysis Speech synthesis Speech coding Nonlinear filters Stochastic processes Low pass filters linear predictive coding Frequency Feedback loop Quantization

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On time alignment and metric algorithms for speech recognition

On time alignment and metric algorithms for speech recogniti...

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International Conference on Information Intelligence and Systems

作者： E.A. Yfantis T. Lazarakis A. Angelopoulos J.D. Elison Y. Zhang Image Processing Laboratory Computer Science Department University of Nevada Las Vegas USA Satrad Las Vegas NV USA

An algorithm for comparing speech waveforms to decide if the spoken utterance is part of a given vocabulary of word waveforms or not, and if it is part of the vocabulary, to choose the matching word is presented. Our algorithm has been implemented in connection with our own vector interpolation alignment algorithm which is faster than dynamic time warping and yet as accurate as dynamic time warping. This vector interpolation, has a classification rate comparable to that of dynamic time warping. While vector interpolation is able to match dynamic time warping for recognition accuracy, it requires significantly less computation, making it much faster than DTW based algorithms. Both algorithms are presented and a comparison of the two is made. Also an alternative algorithm, where the number of intervals of the two utterances to be compared is the same, where the length of the intervals in one utterance is different than the length of the intervals in the other utterance, has been investigated. When appropriate adjustments are made so that the beginning and end of the two utterances match, this algorithm has a classification rate comparable to that of dynamic time warping. Furthermore an alternative to LPC analysis for utterance recognition is presented. Unlike LPC which is an extrapolation algorithm, our algorithm is an interpolation algorithm. Theoretically our algorithm has smaller variance and smaller mean square error than the LPC algorithm. Preliminary results show that our algorithm provides high probability of correct classification.

关键词： Speech recognition linear predictive coding Interpolation Automatic speech recognition Speech processing Vocabulary Extrapolation Mean square error methods Information retrieval Dynamic programming

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Speech recognition using fuzzy logic

Speech recognition using fuzzy logic

引用

International Joint Conference on Neural Networks (IJCNN)

作者： Yong-Qian Ying Peng-Yung Woo Electrical Engineering Department Northern Illinois University Dekalb USA

Speech recognition is one of the important areas in digital speech processing. The study of speech recognition is a part of a quest for artificially intelligent machines that can hear and understand spoken information. The conventional methods for speech recognition are very complicated and time consuming. To apply fuzzy logic to speech recognition is a new attempt in digital speech processing. The approach proposed in the paper simplifies the algorithm in speech recognition and makes the real-time processing time shorter. The situation considered in this paper is the simplest, i.e., the situation of speaker dependence, small vocabulary and isolated words.

关键词： Speech recognition Fuzzy logic linear predictive coding Speech processing Testing Hidden Markov models Autocorrelation Artificial intelligence Machine intelligence Vocabulary

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Quantization of SEW and REW components for 3.6 kbit/s coding based on PWI

Quantization of SEW and REW components for 3.6 kbit/s coding...

引用

IEEE Workshop on Speech coding Proceedings

作者： U. Bhaskar S. Nandkumar K. Swaminathan G. Zakaria Hughes Network Systems Germantown MD USA

The design of a prototype waveform interpolation (PWI) based codec, operating at 3.6 kbit/s, is presented with main focus on the quantization of the slowly evolving waveform (SEW) and rapidly evolving waveform (REW) components. The SEW magnitude component is quantized using a hierarchical mean-shape-gain predictive vector quantization approach. SEW phase is derived using a phase model, based on a measure of voice periodicity. The REW magnitude is quantized using a gain and a sub-band based shape. The REW phase is obtained by high pass filtering a weighted combination of the SEW and a white noise process.

关键词： Frequency Prototypes Vector quantization Speech Interpolation linear predictive coding Band pass filters Low pass filters Codecs Phase measurement

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An adaptive post-filtering technique based on the modified Yule-Walker filter

An adaptive post-filtering technique based on the modified Y...

引用

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

作者： A. Mustapha S. Yeldener COMSAT Laboratories Clarksburg MD USA

This paper presents an adaptive time-domain post-filtering technique based on the modified Yule-Walker filter. Conventionally, post-filtering is derived from an original LPC spectrum. In general, this time-domain technique produces unpredictable spectral tilt that is hard to control by the modified LPC synthesis, inverse and high pass filtering and causes unnecessary attenuation or amplification of some frequency components that introduces muffling in speech quality. This effect increases when voice coders are tandemed together. Another approach of designing a post-filter was developed by McAulay and Quatieri (1991) which can only be used in sinusoidal based speech coders. We have also developed another new time-domain post-filtering technique. This technique eliminates the problem of spectral tilt in the speech spectrum and can be applied to various speech coders. The new post-filter has a flat frequency response at the formant peaks of the speech spectrum. Instead of looking at the modified LPC synthesis, inverse, and high pass filtering in the conventional time-domain technique, we gather information about the poles of the LPC spectrum in the new technique. This post-filtering technique has been used in a 4 kb/s harmonic excitation linear predictive coder (HE-LPC) and a subjective listening tests have indicated that this technique outperforms the conventional one in both one and two tandem connections.

关键词： Time domain analysis linear predictive coding Speech synthesis Adaptive filters Attenuation Frequency synthesizers Frequency response Information filtering Information filters Power harmonic filters

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Split band LPC based adaptive multi-rate GSM candidate

Split band LPC based adaptive multi-rate GSM candidate

引用

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

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

The European Telecommunications Standards Institute (ETSI) has launched a competition for a new mobile communications standard designed to provide better performance than the current GSM standard. This standard is to be called AMR for adaptive multi-rate: the source and channel coding rates can be adapted depending on the state of the channel, thus providing optimal balance between them at any time. The University of Surrey has submitted a candidate for this competition through the Mobile VCE. This candidate was the only one amongst eleven to use a vocoder in the half-rate GSM channel instead of a CELP based coder. The testing which took place as part of the first stage of the competition has shown that this candidate was among the best. This paper presents the system submitted for the half-rate channel as well as the results of the testing.

关键词： linear predictive coding GSM Vocoders Telecommunication standards Background noise Testing Communication standards Channel coding Speech Qualifications

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Investigation of a least squares method for high order filter synthesis

Investigation of a least squares method for high order filte...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M. Price M. Sandler Department of Electronic Engineering King''s College London UK

This paper defines a least squares algorithm for synthesising the Parallel Interconnected Subfilters (PIS) class of high order filter for speech codec applications. An investigation of the performance of the algorithm... 详细信息

关键词： Least squares methods Filters linear predictive coding Partitioning algorithms Decoding Polynomials Speech synthesis Educational institutions Speech codecs Speech coding

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Comparing audio compression using wavelets with other audio compression schemes

Comparing audio compression using wavelets with other audio ...

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Canadian Conference on Electrical and Computer Engineering (CCECE)

作者： E.-B. Fgee W.J. Phillips W. Robertson Department of Electrical and Computer Engineering DalTech Dalhousie University Halifax NS Canada

Speech compression is the technology of converting human speech into an efficient encoded representation that can be decoded to produce a close approximation of the original signal. In this paper, we propose a new algorithm which compresses speech signals using a wavelet compression technique. The performance of this method is compared against the following representative coding and compression schemes: adaptive differential pulse code modulation (ADPCM) which reduces the transmitted data by a factor of two; linear predictive coding (LPC) with compression ratio of more than twelve to one; linear predictive coding algorithm using the United States Department of Defense Standard 1015 with compression ratio of 26:1; Global System Mobile (GSM) algorithm which reduces the transmitted data by a factor of five. The following parameters are compared: (i) quality of the reconstructed signal after decoding; (ii) compression ratios. (iii) signal to noise ratio (SNR); (iv) peak signal to noise ratio (PSNR); (v) normalized root mean square error (NRMSE).

关键词： Audio compression Speech linear predictive coding Decoding Pulse modulation Signal to noise ratio PSNR Humans Pulse compression methods Modulation coding

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Text-dependent speaker identification using LPC and DTW for Thai language

Text-dependent speaker identification using LPC and DTW for ...

引用

IEEE Region 10 International Conference TENCON

作者： C. Wutiwiwatchai V. Achariyakulporn C. Tanprasert Software and Language Engineering Laboratory National Electronics and Computer Technology Center Nation Science and Technology Development Agency Ministry of Science Technology and Environment Thailand Bangkok Thailand

This paper proposes a text-dependent speaker identification system applied to Thai language. Isolated digits 0-9 and their concatenations are used for speaking text. linear prediction coefficients (LPC) are extracted and formed as feature vectors represented each speech signal. Dynamic time warping (DTW) is used to measure distances between referenced and evaluated vectors. These distances, indicating nearness of unknown vectors to references, incorporated with the K-nearest neighbor (KNN) decision technique are used to decide who possesses those unknown vectors. The experimental results have shown that the best identification rate for a single digit is 95.83% and the highest rate for concatenated digits of top-3, top-5, and top-7 are 98.75%, 100%, and 99.20%, respectively.

关键词： linear predictive coding Natural languages Speaker recognition Feature extraction Artificial neural networks Vectors Loudspeakers Isolation technology Pattern matching Hidden Markov models

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Noise immune speech recognition system

Noise immune speech recognition system

引用

National Conference on Radio Science (NSRC)

作者： M. Gadallah E. Soleit A. Mahran Kobry El-koppa Military Technical College Cairo Egypt

This paper investigates the performance of an isolated word recognition (IWR) system in a noisy environment. Two approaches have been demonstrated to overcome the effect of the noise on the recognition accuracy. These approaches are, using noise immune features and reference model contamination. The performance is evaluated in a noisy environment at different signal-to-noise ratios (SNR), with different feature extraction techniques including linear predictive coding (LPC), cepstrum analysis, weighted cepstrum analysis, and perceptual linear predictive coding (PLP). The performance of these features is compared based on the recognition accuracy. The results have shown that the PLP features exhibits the best noise immunity and recognition accuracy among the studied features.

关键词： Speech recognition Working environment noise Signal to noise ratio linear predictive coding Cepstrum Cepstral analysis Signal analysis Performance analysis Contamination Feature extraction

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