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检索条件"主题词=linear predictive coding"

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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...

引用

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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New algorithm for spectral smoothing and envelope modification for LP-PSOLA synthesis

New algorithm for spectral smoothing and envelope modificati...

引用

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

作者： F.M. Gimenez de los Galanes M.H. Savoji J.M. Pardo Dpto. Ingeniería Electrónica U. Politécnica de Madrid. E.T.S.I.Telecommunicatión Ciudad Universitaria Madrid Spain Dpto. Electrónica University de Castilla-La Mancha Santander Spain

The final quality of a concatenation synthesis system is directly related to the continuity of the spectrum at the concatenation point. Due to the subjective auditory masking, if we minimize the spectral distortion in the formant frequencies, the quality will increase significantly. We present, along with results concerning pitch marking, an algorithm capable of modifying the LPC envelope in a flexible way which is the heart of a spectral smoothing module for a diphone-based linear prediction pitch-synchronous overlap-add (LP-PSOLA) concatenation system.< >

关键词： Smoothing methods linear predictive coding Signal synthesis Speech synthesis Distortion Signal processing Filtering Heart Frequency measurement Time measurement

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Active power minimization of a sound source in a reverberant space

Active power minimization of a sound source in a reverberant...

引用

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

作者： M. Tohyama A. Suzuki K. Sugiyama Human Interface Lab. NTT Tokyo Japan NTT Human Interface Laboratories Musashino Tokyo Japan

Sound power output from a source can be reduced in a closed space by using secondary sources. The minimum power response (MPR) from sound sources is analyzed based on the modal theory. It is shown that the MPR can be approximately described using the modal overlap properties. The power reduction is confirmed experimentally for a pure tone source in a large reverberation room. The corner method for sensor location is also discussed.< >

关键词： Frequency Resonance Reverberation Acoustic sensors Wave functions Humans Laboratories Minimization methods Position measurement linear predictive coding

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Voice operated intelligent wheelchair - VOIC

Voice operated intelligent wheelchair - VOIC

引用

IEEE International Symposium on Industrial Electronics (ISIE)

作者： G. Pacnik K. Benkic B. Brecko Faculty of Electrical Engineering and Computer Science Institute of Robotics Maribor Slovenia

A development of the intelligent wheelchair lab prototype is shown in the paper. VOIC is designed for physically disabled person, who can not control their movements and control the wheelchair with the joystick. The article describes basic components of voice recognition and wheelchair control system. Voice recognition begins with input signal sampling, word isolation, LPC cepstral analysis, coefficient dimension reduction and trajectory recognition using fixed point approach with neural networks. Wheelchair control system is divided into system for sensors data acquisition and system for wheelchair steering. The complexity of the voice recognition is reduced using LPC cepstral analysis and coefficient dimension reduction with minimum loss of vital information. A proper decision for the speech recognition using neural networks is supported with experimental results.

关键词： Wheelchairs Speech recognition Control systems linear predictive coding Cepstral analysis Neural networks Prototypes Signal sampling Sensor systems Data acquisition

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Generalized autoregressive spectral estimation

Generalized autoregressive spectral estimation

引用

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

作者： J. Tsao W.-J. Shyu Department of Electrical Engineering National Taiwan University Taipei Taiwan

An autoregressive spectral estimation method is developed to reduce the noise effect in prediction coefficient estimation. This method solves the prediction coefficients from a generalized Yule-Walker equation which is formed by the data and its generalized autocorrelation sequence. This method provides several control parameters for the spectral estimator to combat the unmodeled additive noise in the linear least square sense. Through the efficient use of information by this method, data size will be directly helpful in noise suppression.< >

关键词： Autocorrelation linear predictive coding Noise reduction Least squares approximation Difference equations Random processes Additive noise Spectral analysis White noise Councils

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The effects of subtractive-type speech enhancement/noise reduction algorithms on parameter estimation for improved recognition and coding in high noise environments

The effects of subtractive-type speech enhancement/noise red...

引用

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

作者： W.M. Kushner V. Goncharoff C. Wu V. Nguyen J.N. Damoulakis Ocean Systems Division Martin Marietta Aero and Naval Systems Glen Burnie MD USA

The authors present the results of a study designed to investigate the effects of subtractive-type noise reduction algorithms on LPC-based spectral parameter estimation as related to the performance of speech processors operating with input SNRs of 15 dB and below. Subtractive noise preprocessing greatly improves the SNR, but system performance improvement is not commensurate. LPC spectral estimation is affected by the character of the residual noise which exhibits greater variance and spectral granularity than the original broadband noise. The study shows that removing less than the full amount of noise and whitening it improves spectral estimation and speech device performance. Techniques and performance results are presented.< >

关键词： Speech enhancement Noise reduction Parameter estimation Acoustic noise Signal to noise ratio linear predictive coding Speech processing Speech recognition Cepstral analysis Speech coding

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What Influence Does the Type of Extracted Audio Features Have on Emotional States?

What Influence Does the Type of Extracted Audio Features Hav...

引用

IEEE International Conference on Automation, Quality and Testing, Robotics, AQTR

作者： Toma Telembici Lorena Muscar Corneliu Rusu Signal Processing Group Faculty of Electronics Telecommunications and Information Technology Technical University of cluj-Napoca Cluj-Napoca Romania

ISBN: (数字)9798350361933

ISBN: (纸本)9798350361940

This paper presents the way in which different audio emotional states are classified using various feature extraction types. Our audio soundbase for robot service application, in Romanian language, alongside with four established soundbases, Crema, Ravdess, Savee and Emo-DB were used in this paper. We analyzed the influence of different feature extraction methods upon classification accuracy. We focus on Mel Frequency Cepstral Coefficients (MFCC), linear predictive coding (LPC), Magnitude-based Spectral Root Cepstral Coefficients (MSRCC), Normalized Gammachirp Cepstral Coefficients (NGCC) and Gammatone Frequency Cepstral Coefficients (GFCC). One classifier, k-nearest neighbor (k-NN), is utilized to evaluate the effectiveness of each feature set in recognizing emotional states. We compared the correct classification rates to determine which features have the best results in the emotional recognition field. We show that by using $30-\mathrm{NGCC}$ and 5 -nearest neighbors we reach an overall correct classification rate of $96.46 \%$ in the testing phase. Computational time was also incorporated into the analysis, including the testing and training phases. This study makes contributions to the field of emotion recognition from audio signals, facilitating advancements in human-computer interaction.

关键词： Training Human computer interaction Emotion recognition Automation Accuracy Feature extraction linear predictive coding

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A two-step speech compression system with vector quantizing

A two-step speech compression system with vector quantizing

引用

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

作者： A. Buzo A. Gray R. Gray J. Markel Signal Technology Inc. SantaBarbara CA Information Systems Laboratory University of Stanford Stanford CA USA

A training sequence of speech data is used to design a two-step speech compression system, based upon either single speakers or multiple speakers. The system is designed to minimize an average spectral distortion over the training sequence, leading to an identification step using linear prediction techniques followed by a vector quantizer. The system is then used to compress test sequences of speech data, leading to much lower bit rates than obtained using scalar quantization for equivalent distortions. For the same numerical distortion, 20-bits/frame were required using "optimal" scalar bit allocation and quantization, whereas 8-bits/frame were required using vector quantization. Results are presented in the form of numerical distortion measures and analog tapes of synthesized speech.

关键词： Distortion measurement Switches Speech processing Quantization linear predictive coding Random processes Noise measurement Vectors Bit rate Speech synthesis

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Evaluation of the Limiting Permissible Concentration Concept in Ocean Dumping Assessments

Evaluation of the Limiting Permissible Concentration Concept...

引用

OCEANS

作者： J. Fava W. McCulloch J. Gift W. Rue EA Engineering Science and Technology Inc. Sparks MD USA

The U.S. Ocean Dumping Regulations and Criteria (40 CFR 220-229) limit the types, amount, and release rates of waste into the marine environment. Materials to be considered for ocean dumping must be assessed by a technical evaluation of their potential environmental impact. Based on bioassay results, a limiting permissible concentration (LPC) is established that shall not be exceeded beyond the disposal site boundaries during initial mixing, and shall not be exceeded at any point in the marine environment after initial mixing. The LPC determination is made up of three critical variables: initial mixing considerations, acute toxicity threshold, and the use of an application factor to protect against chronic effects. Thus, the LPC is not a constant value, but is affected by laboratory results and dumping practices.

关键词： Oceans linear predictive coding Protection Biological materials Safety Organisms Testing Marine technology Materials science and technology Sparks

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On reducing the buzz in LPC synthesis

On reducing the buzz in LPC synthesis

引用

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

作者： M. Sambur A. Rosenberg L. Rabiner C. McGonegal Bell Laboratories Murray Hill NJ USA

A method for reducing the characteristic buzz from LPC synthetic speech is presented. The method consists of the use of an non-impulse source for exciting the LPC synthesizer during voiced sounds. One novel feature is that the temporal parameters of the source are kept in fixed proportion to the pitch period. An extensive perceptual experiment has shown that the resulting quality of the synthesis is significantly preferred over the quality of the standard LPC synthesis.

关键词： linear predictive coding Speech synthesis Synthesizers Interpolation predictive models Speech analysis Poles and zeros Speech processing Ear System testing

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