IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '78, Tulsa, Oklahoma, USA, April 10-12, 1978. IEEE, 1978. [doi]

Conference: icassp1978

Abstract is missing.

- Studies on pattern recognition approach to voiced-unvoiced-silence classification1-4 [doi]
- Improvement of voicing decisions by use of context5-7 [doi]
- Epoch extraction from linear prediction residual8-11 [doi]
- Maximum likelihood pitch estimation using state-variable techniques12-14 [doi]
- Interactive digital inverse filtering and its relation to linear prediction methods15-18 [doi]
- A new method of Cepstrum analysis by using comb lifter19-22 [doi]
- Arma modeling applied to linear prediction of speech23-26 [doi]
- Study of an adaptive lattice structure for linear prediction analysis of speech27-30 [doi]
- Real-zeros in pitch detection31-34 [doi]
- Identification of nonstationary components by complex spectral zero analysis35-38 [doi]
- Two-dimensional processing of spectrogram data39-42 [doi]
- A high-quality all-digital sound spectrograph developed for speech signal analysis43-46 [doi]
- Realization of digital transfer functions using cascaded normalized two-pairs47-50 [doi]
- Calculation of all equivalent and canonic 2nd order digital filter structures51-54 [doi]
- State-structures and minimal state-structures for arbitrary digital filters55-57 [doi]
- Realization of a class of two-dimensional analog ladders with applications to wave digital filters58-61 [doi]
- A figure of merit for digital filters62-66 [doi]
- Formats in audio bandwidth processing applications67-70 [doi]
- Digital filter realizations without overflow oscillations71-74 [doi]
- An approach to eliminate roundoff errors in digital filters75-78 [doi]
- Selection of data windows for digital signal processing79-82 [doi]
- Adaptive lattice methods for linear prediction83-86 [doi]
- An adaptive lattice structure for noise-cancelling applications87-90 [doi]
- Non-optimal convergence of adaptive LMS with uncorrelated data91-95 [doi]
- Selecting the length of an adaptive transversal filter96-99 [doi]
- An adaptive equalizer with significantly reduced number of operations100-104 [doi]
- Real-time signal processing for unbiased system identification105-108 [doi]
- Adaptive data orthogonalization109-112 [doi]
- Some mathematical results on the effects on digital adaptive filters of implementation errors and noise113-117 [doi]
- Simple adaptive IIR filtering118-122 [doi]
- A short-term sequential regression algorithm123-126 [doi]
- On the maximum achievable conversion efficiency on a parametric acoustic array127-129 [doi]
- Vibration sensitivity of the parametric acoustic receiving array130-133 [doi]
- Errors in array response calculations due to incorrectly folding the vertical acoustic arrival structure into beam pattern134-136 [doi]
- Control of the directional response of acoustic transducer array elements on curved surfaces by modification of the diffracted field137-140 [doi]
- Array processing using the frequency-wavenumber approach141-142 [doi]
- Effect of sampling errors on array gain143-147 [doi]
- Synthetic array processing for underwater mapping applications148-151 [doi]
- A flexible towed sonar for ocean acoustic measurements152-154 [doi]
- An audio response method for CAI services155-158 [doi]
- Speech synthesis using real-time software159-162 [doi]
- A mixed-source model for speech compression and synthesis163-166 [doi]
- Implementation of a channel vocoder synthesizer using a fast, time-multiplexed digital filter167-170 [doi]
- An excitation function for LPC synthesis which retains the human Glottal phase characteristics171-174 [doi]
- Votrax real time hardware for phoneme synthesis of speech175-178 [doi]
- Speech generation through waveform synthesis179-182 [doi]
- Comparisons of system identification methods in the presence of high noise levels and bandlimited inputs183-187 [doi]
- Linear prediction and maximum entropy spectral analysis of finite bandwidth signals in noise188-191 [doi]
- Comparative study of iterative deconvolution algorithms192-195 [doi]
- Image restoration of space variant blurs by sectioned methods196-198 [doi]
- Nonlinear filter for inversion of channel distortion199-202 [doi]
- Observations on linear estimation203-207 [doi]
- New stochastic realization algorithms for identification of ARMA models208-213 [doi]
- Analysis and representation of composite signals by cepstral inverse filtering214-217 [doi]
- Multilevel crossing rates for automated signal classification218-222 [doi]
- Target identification from radar signatures223-227 [doi]
- Multiband FIR digital filter design algorithm for radar clutter suppression228-231 [doi]
- Typical performance characteristics of a two-dimensional CFAR232-235 [doi]
- A technique for pole-zero modeling of complex-valued autocorrelations236 [doi]
- A programmable sonar signal processor237-240 [doi]
- Reconnaissance sonar for deep ocean seamount detection241-244 [doi]
- An at-sea system for the prediction of underwater sound propagation245-247 [doi]
- Inherent errors in sonar range prediction248-251 [doi]
- Design considerations for feedback amplifiers252-254 [doi]
- Audibility of transient intermodulation distortion255-262 [doi]
- Omitted factors in audio circuit design263-266 [doi]
- Computer analysis of transient distortion and low transient distortion amplifier design267-269 [doi]
- Electroacoustic distortions: Multidimensional analysis of hearing aid transduced speech and music270-274 [doi]
- A comparison between two approaches to automatic speaker recognition275-278 [doi]
- Extraction of speaker-specific features from spoken code sentences279-282 [doi]
- New techniques for text-independent speaker identification283-286 [doi]
- Text-independent speaker identification from a large linguistically unconstrained time-spaced data base287-290 [doi]
- Text-independent speaker identification based on piecewise canonical discriminant analysis291-294 [doi]
- Automatic speaker identification for a large number of speakers295-298 [doi]
- 2400/16, 000 Bps Multirate voice processor299-302 [doi]
- High quality adaptive predictive coding of speech303-306 [doi]
- 9.6/7.2 Kbps Voice excited predictive coder (VEPC)307-311 [doi]
- An optimal adaptation logic for delta modulation312-315 [doi]
- Forward-adaptive delta modulator without explicit transmission of step size316-319 [doi]
- 32 Kbps CCITT Compatible split band coding scheme320-325 [doi]
- Evaluation of LPC/CVSD tandem connections326-329 [doi]
- A spectral enhancement procedure for the wideband/Narrowband tandem330-333 [doi]
- Towards a variable frame rate speech transmission system with frame selection by time-domain segmentation - a status report334-337 [doi]
- Two dimensional data compression of speech338-340 [doi]
- Two-dimensional speech compression341-344 [doi]
- High resolution autoregressive spectrum analysis using noise power cancellation345-348 [doi]
- High resolution spectral analysis of sinusoids in correlated noise349-351 [doi]
- Frequency estimation by linear prediction352-356 [doi]
- Improvement of autoregressive spectral estimates in the presence of noise357-360 [doi]
- The accuracy of center frequency estimators using linear predictive methods361-364 [doi]
- Complex covariance/Maximum entropy doppler estimates for pulsed CO2lidar365-368 [doi]
- An error formula for iterative prefiltering frequency estimates369-371 [doi]
- Extrapolation and spectral estimation for bandlimited signals372-374 [doi]
- Constant-Q signal analysis and synthesis375-378 [doi]
- Spectrum analysis using frequency-domain adaptive windowing379-382 [doi]
- Spectrum estimation of non-uniform sampled data383-386 [doi]
- Input signals for sampling and identification of ecological systems387-390 [doi]
- Input, signals and control in ecosystems391-397 [doi]
- Extension of Levins loop analysis to transient and periodic disturbances398-401 [doi]
- Estimating Oxygen demand in aquatic ecosystems402-404 [doi]
- Gaps in the technology of speech understanding405-408 [doi]
- Computing relative redundancy to measure grammatical constraint in speech recognition tasks409-412 [doi]
- Dynamic programming, the viterbi algorithm, and low cost speech recognition413-417 [doi]
- Automatic recognition of continuously spoken sentences from a finite state grammer418-421 [doi]
- Recognition of continuously read natural corpus422-424 [doi]
- A voice-input programming system using basic-like language425-428 [doi]
- The Dawid speech recognition system429-432 [doi]
- Fuzzy semantic network for a speech understanding system - an experimental study433-436 [doi]
- A simple vocoder437-440 [doi]
- Low bit rate cepstral vocoder using the log magnitude approximation filter441-444 [doi]
- CCD CZT Spectral analysis applied to real time homomorphics speech analysis-synthesis445-449 [doi]
- Implementation of a pole-zero analysis - synthesis system for speech450-453 [doi]
- A variable frame length linear predictive coder454-457 [doi]
- A method for reducing the transmission rate of a channel vocoder by using frame interpolation458-461 [doi]
- Providing channel error protection for a 2400 bps linear predictive coded voice system462-465 [doi]
- Channel coding considerations for digital speech encoded by linear prediction466-471 [doi]
- Linear predictive coding of speech signals in a high ambient noise environment472-475 [doi]
- Linear prediction techniques in the Walsh spectral domain for speech analysis and synthesis476-479 [doi]
- Application of canonical coordinate methods to the characterization of a family of error minimizing signal compression techniques480-482 [doi]
- An estimate of the order of an optimal FIR band-pass digital filter483-486 [doi]
- Digital filters with prescribed zeros487-490 [doi]
- Synthesis of digital filters with very low sensitivity of the frequency response to the change of the cofficients491-494 [doi]
- Design of IIR filters using Pseudo-Boolean methods495-498 [doi]
- Design and applications of uniform digital bandpass filter banks499-503 [doi]
- A multiplexing scheme for multirate digital filtering with half-band filters504-507 [doi]
- New class of recursive digital filters for decimation508-511 [doi]
- An optimal filter design for variable sampling rates512-515 [doi]
- Digital lowpass filtering using the discrete Hilbert transform516-519 [doi]
- Nonstationary signal processing and model validation520-523 [doi]
- Recursive derivation of reflection coefficients from noisy seismic data524-528 [doi]
- On digital signal modelling and classification with the teleseismic data529-531 [doi]
- Acoustic wave propagation by finite-difference techniques532 [doi]
- Ultrasonic measurements of defects in metals using cepstral processing533-537 [doi]
- Digital recording of ultrasonic signals538-540 [doi]
- A comparison of linear prediction, FFT, and zero-crossing analysis techniques for vowel recognition541-545 [doi]
- Speaker-independent vowel indetification in continuous speech546-548 [doi]
- A probabalistic vector model for identification of intervocalic stop consonants549-552 [doi]
- A comparative study of phonemic recognition by discrete orthogonal transforms553-556 [doi]
- A phoneme recognition system based on human audition557-560 [doi]
- Application of the subspace method to speech recognition561-564 [doi]
- Application of novelty filter to segmentation of speech565-568 [doi]
- Signal/Quantizing-distortion ratio measurements of fast-adaptive delta modulation systems569-572 [doi]
- Predictive coding of speech signals and subjective error criteria573-576 [doi]
- Speech synthesis by linear interpolation of spectral parameters between dyad boundaries577-580 [doi]
- Perceptual and objective evaluation of speech processed by adaptive differential PCM581-585 [doi]
- A study of complexity and quality of speech waveform coders586-590 [doi]
- Objective speech quality evaluation of narrowband LPC vocoders591-594 [doi]
- Statistical correlation between objective and subjective measures for speech quality595-598 [doi]
- Estimation of LPC coefficients from speech waveforms degraded by additive random noise599-601 [doi]
- An investigation of several frequency-domain processing methods for enhancing the intelligibility of speech in wideband random noise602-605 [doi]
- Suppression of noise in speech using the saber method606-609 [doi]
- LMS Adaptive filtering for enhancing the quality of noisy speech610-613 [doi]
- Two dimensional filtering using fermat number transforms614-618 [doi]
- Signal processing with number theoretic transforms and limited word lengths619-623 [doi]
- q+ 1624-627 [doi]
- Simulation of large length filters using fermat number transforms628-631 [doi]
- An inplace self recordering FFT632-633 [doi]
- Real-factor FFT algorithms634-637 [doi]
- New algorithms for convolution and DFT based on polynomial transforms638-641 [doi]
- M-Adic invariant filters642-645 [doi]
- Reconstruction of signals from their linear mapping image646-650 [doi]
- Linear detection filtering for the context of a least-squares estimator for signal processing applications651-654 [doi]
- Dealiasing of the spectra of sampled noise655-658 [doi]
- Underwater sound arrival angle estimation by multiple cross-correlation measurements659-664 [doi]
- The least squares estimation of time delay and its use in signal detection665-669 [doi]
- Confidence bounds for magnitude-squared coherence estimates670-673 [doi]
- An all-digital phase-measurement techniques using clipped-quadrature correlation674-677 [doi]
- Detection performance of the smooth coherence transform (SCOT)678-683 [doi]
- Object classification using sonar data684-687 [doi]
- A study of stochastic processes associated with sonar detection688-691 [doi]
- Acoustic transfer function of the ocean for a motional source692-695 [doi]
- Display and interpretation of a time-spread underwater acoustic channel's bandlimited impulse response696-699 [doi]
- Some experiments with a syntax directed speech recognition system700-703 [doi]
- Performance analysis of syntactic-recognizer of isolated words704-707 [doi]
- Recognition of monosyllabic words in continuous sentences using composite word templates708-711 [doi]
- An automatic word spotting system for conversational speech712-717 [doi]
- Directory assistance by means of automatic recognition of spoken spelled names718-721 [doi]
- An approach to speech recognition using syllabic decision units722-725 [doi]
- A word recognition method from a classified phoneme string in the Lithan speech understanding system726-730 [doi]
- Nearest neighbour decision rule for vowel and digit recognition731-734 [doi]
- Spoken word recognition system for unlimited speakers735-738 [doi]
- Two-dimensional signal processing from hexagonal rasters739-742 [doi]
- Digital decomposition and representation of video signals using projection theory743-746 [doi]
- On generation of two-dimensional data747-750 [doi]
- Inversion of block - Toeplitz matrices using bivariate szego polynomials751-752 [doi]
- Two-dimensional zoom FFT753-756 [doi]
- Recursive digital filters in image processing757-760 [doi]
- Design of stable 2-D half-plane recursive filters using spectral factorization761-764 [doi]
- Design of inherently stable two-dimensional recursive filters imitating the behaviour of one-dimensional analog filters765-768 [doi]
- An algorithm for testing stability of two-dimensional digital recursive filters769-772 [doi]
- Computer aided generation of two dimensional transfer functions from one dimensional transfer functions773-776 [doi]
- Design of semicausal two-dimensional recursive filters777-781 [doi]
- A well suited two-dimensional linear recursive filter for image processing782-787 [doi]
- A simple hardware implementation of digital notch filters788-791 [doi]
- Digital phase locked loop792-795 [doi]
- Parallel counter design using four-valued threshold logic796-799 [doi]
- An error anaylsis of a FFT implementation using the residue number system800-803 [doi]
- Techniques for residue-to-analog conversion for high data rate digital filtering804-807 [doi]
- 20Channel 300 MHz bandwidth digital spectrum analyzer808-811 [doi]
- A 16-bit microprocessor-based digital filter architecture812-815 [doi]
- The compass block-diagram compiler: A new development in signal-processor programming816-819 [doi]
- Impulse response testing of acoustic spaces820-823 [doi]
- Super directive spectrum analyzer by use of moving microphones824-827 [doi]
- Free-field measurements for a loudspeaker system in a normal room-Using digital signal processing techniques828-831 [doi]
- Correction of near-field acoustic measurements made with arbitrary measuring transducers832-835 [doi]
- Reflection of spherical sound wave from a rigid sphere836-839 [doi]
- Statistical properties of the poisson reverberation envelope840-845 [doi]
- On an extension of A. N. Krylov's numerical method for determining the frequencies of small vibrations of systems with damping846-847 [doi]