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Iterative techniques for minimum phase signal reconstruction from phase or magnitude

December 1, 1981
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Journal Article
Author:
Thomas F. Quatieri
…
Published in:
IEEE Trans. on Acoustics, Speech & Signal Processing, Vol. ASSP-29, No.6, Dec. 1981, pp.1187-1193.
Topic:
speech enhancement
R&D area:
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Summary

In this paper, we develop iterative algorithms for reconstructing a minimum phase sequence from pthhea se or magnitude of its Fourier transform. These iterative solutions involve repeatedly imposing a causality constraint in the time domain and incorporating the known phase or magnitude function in the frequency domain. This approach is the basis of a new means of computing the Hilbert transform of the log-magnitude or phase of the Fourier transform of a minimum phase sequence which does not require phase unwrapping. Finally, we discuss the potential use of this iterative computation in determining samples of the unwrapped phase of a mixed phase sequence.
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Summary

In this paper, we develop iterative algorithms for reconstructing a minimum phase sequence from pthhea se or magnitude of its Fourier transform. These iterative solutions involve repeatedly imposing a causality constraint in the time domain and incorporating the known phase or magnitude function in the frequency domain. This approach is...

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Iterative techniques for minimum phase signal reconstruction from phase or magnitude

Recursive two-dimensional signal reconstruction from linear system input and output magnitudes

May 1, 1981
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Journal Article
Author:
Thomas F. Quatieri
Published in:
Proc. IEEE, Vol. 69, No. 5, May 1981, pp. 667-668.
Topic:
signal processing
R&D area:
R&D group:

Summary

A recursive algorithm is presented for reconstructing a two-dimensional complex signal from its magnitude and the magnitude of the output of a known linear shift-invariant system whose input is the desired signal. The recursion has a simple geometric interpretation, and is easily extended to causal, shift-varying systems.
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Summary

A recursive algorithm is presented for reconstructing a two-dimensional complex signal from its magnitude and the magnitude of the output of a known linear shift-invariant system whose input is the desired signal. The recursion has a simple geometric interpretation, and is easily extended to causal, shift-varying systems.

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Recursive two-dimensional signal reconstruction from linear system input and output magnitudes

Voice communication in integrated digital voice and data networks

September 1, 1980
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Journal Article
Author:
Theodore Bially
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Published in:
IEEE Trans. Commun., Vol. COM-28, No. 9, September 1980, pp. 1478-90.
Topic:
signal processing
R&D area:
R&D group:

Summary

Voice communication networks have traditionally been designed to provide either analog signal paths or fixed-rate synchronous digital connections between individual subscribers. These designs were aimed at accommodating the "streamlike" character of speech, which has traditionally been considered to flow from source to destination at a more or less constant rate. By way of contrast, interactive and computer-to-computer data transactions tend to be "bursty" in nature, and this has given rise to the development of packet-switching methods for data communications. The dichotomous nature of these two major traffic classes and the apparent conflict between the types of network services they require has resulted in the deployment of separate military communications facilities for voice and data. A challenge in the design of future systems is to achieve overall economy and flexibility in the allocation of resources via the efficient integration of both traffic types in common network facilities. This paper summarizes a number of advanced concepts for switching and flow control of combined voice and data traffic in integrated environments. Performance characteristics are described based on analysis results and computer simulation studies for both multilink terrestrial and broadcast satellite network topologies.
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Summary

Voice communication networks have traditionally been designed to provide either analog signal paths or fixed-rate synchronous digital connections between individual subscribers. These designs were aimed at accommodating the "streamlike" character of speech, which has traditionally been considered to flow from source to destination at a more or less constant rate...

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Voice communication in integrated digital voice and data networks

Convergence of iterative nonexpansive signal reconstruction algorithms

August 1, 1980
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Journal Article
Author:
Victor T. Tom
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Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-29, No. 5, October 1981, pp. 1052-1059.
Topic:
signal processing
R&D area:
R&D group:

Summary

Iterative algorithms for signal reconstruction from partial time- and frequency-domain knowledge have proven useful in a number of application areas. In this paper, a general convergence proof, applicable to a general class of such iterative reconstruction algorithms, is presented. The proof relies on the concept of a nonexpansive mapping in both the time and frequency domains. Two examples studied in detail are time-limited extrapolation (equivalently, band-limited extrapolation) and phase-only signal reconstruction. The proof of convergence for the phase-only iteration is a new result obtained by this method of proof. The generality of the approach allows the incorporation of nonlinear constraints such as time- (or space-) domain positivity or minimum and maximum value constraints. Finally, the underrelaxed form of these iterations is also shown to converge even when the solution is not guaranteed to be unique.
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Summary

Iterative algorithms for signal reconstruction from partial time- and frequency-domain knowledge have proven useful in a number of application areas. In this paper, a general convergence proof, applicable to a general class of such iterative reconstruction algorithms, is presented. The proof relies on the concept of a nonexpansive mapping in...

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Convergence of iterative nonexpansive signal reconstruction algorithms

Data traffic performance of an integrated circuit and packet-switched multiplex structure

June 1, 1980
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Journal Article
Author:
Marilyn L. Malpass
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Published in:
IEEE Trans. on Commun., Vol. COM-28, No. 6, June 1980, pp. 873-878.
Topic:
signal processing
R&D area:
R&D group:

Summary

Results are developed for data traffic performance in an integrated multiplex structure which includes circuit-switching for voice and packet-switching for data. The results are obtained both through simulation and analysis, and show that excessive data queues and delays will build up under heavy loading conditions. These large data delays occur during periods of time when the voice traffic load through the multiplexer exceeds its statistical average. A variety of flow control mechanisms to reduce data packet delays are investigated. These mechanisms include control of voice bit rate, limitation of the data buffer, and combinations of voice rate and data buffer control. Simulations indicate that these flow control mechanisms provide substantial improvements in system performance.
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Summary

Results are developed for data traffic performance in an integrated multiplex structure which includes circuit-switching for voice and packet-switching for data. The results are obtained both through simulation and analysis, and show that excessive data queues and delays will build up under heavy loading conditions. These large data delays occur...

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Data traffic performance of an integrated circuit and packet-switched multiplex structure

The tradeoff between delay and TASI advantage in a packetized speech multiplexer

November 1, 1979
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Journal Article
Author:
Clifford J. Weinstein
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Published in:
IEEE Trans. on Commun., Vol. COM-27, No. 11, November 1979, pp. 1716-20.
Topic:
signal processing
R&D area:
R&D group:

Summary

A packetized speech multiplexer differs from a circuit-switched TASI system in that the presence of a packet buffer allows a tradeoff where the TASI advantage can be increased at a cost in packet delay. This tradeoff is investigated via a simulation. Results are presented to show the relations between TASI advantage and delay, for both an average delay criterion and a maximum delay criterion. It is shown that, particularly for the case where small numbers of talkers are multiplexed, the packetized system offers significant improvements in TASI advantage over the conventional circuit-switched multiplexer, at modest costs in packet delay.
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Summary

A packetized speech multiplexer differs from a circuit-switched TASI system in that the presence of a packet buffer allows a tradeoff where the TASI advantage can be increased at a cost in packet delay. This tradeoff is investigated via a simulation. Results are presented to show the relations between TASI...

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The tradeoff between delay and TASI advantage in a packetized speech multiplexer

A phrase recognizer using syllable-based acoustic measurements

October 1, 1978
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Journal Article
Author:
Don H. Johnson
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Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-26, No. 5, October 1978, pp. 409-418.
Topic:
speech recognition
R&D area:
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Summary

A system for the recognition of spoken phrases is described. The recognizer assumes that the input utterance contains one of a known set of allowable phrases, which may be spoken within a longer carrier sentence. Analysis is performed on a syllable-by-syllable basis with only the strong syllables considered in the recognition process. Each strong syllable is represented in terms of a set of distinguishing acoustic measurements taken at time points in and around the syllable nucleus. Phrases are represented as sequences of strong syllables. All parameters used in recognition are derived from LPC coefficients. Input speech is limited to 3.3 kHZ upper frequency. Recognition is completed within 1-3 s after the utterance is spoken. An interactive training facility allows flexible composition of key phrase sets. Testing was performed for a number of phrase sets each containing ten or fewer phrases, and included equal numbers of talkers used in training and talkers not used in training. Average phrase recognition accuracy was 95 percent when parameters were derived from unquantized (i.e., 16 bit) LPC coefficients and 90 percent when the LPC coefficients were transmitted to the recognizer across the ARPA network at 3500 bits/s. The recognizer has been incorporated into a user interface system where the parameters required to set up a point-to-point ARPANET voice connection can be established remotely by voice.
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Summary

A system for the recognition of spoken phrases is described. The recognizer assumes that the input utterance contains one of a known set of allowable phrases, which may be spoken within a longer carrier sentence. Analysis is performed on a syllable-by-syllable basis with only the strong syllables considered in the...

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A phrase recognizer using syllable-based acoustic measurements

A linear prediction vocoder with voice excitation

September 29, 1975
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Conference Paper
Author:
Clifford J. Weinstein
Published in:
Proc. EASCON, 29 September - 1 October 1975, pp. 30-a-30-g.
Topic:
speech processing
R&D area:
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Summary

A speech bandwidth compression system, which employs voice excitation in conjunction with a Linear Predictive Coding (LPC) parameterization of the vocal tract filter, is described. To generate the excitation signal, the transmitted speech baseband is broadened at the receiver with a nonlinear distorter, and spectrally flattened by means of an adaptive inverse filter whose parameters are obtained through LPC analysis of the distorted baseband. The voice-excited linear prediction (VELP) system has been implemented in real time on the Fast Digital Processor at Lincoln Laboratory. A detailed description of an 8 kbps version of VELP is given. VELP offers promise as a good quality, medium rate speech compression system which, by avoiding the pitch problem, performs relatively well for telephone quality input speech.
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Summary

A speech bandwidth compression system, which employs voice excitation in conjunction with a Linear Predictive Coding (LPC) parameterization of the vocal tract filter, is described. To generate the excitation signal, the transmitted speech baseband is broadened at the receiver with a nonlinear distorter, and spectrally flattened by means of an...

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A linear prediction vocoder with voice excitation

A system for acoustic-phonetic analysis of continuous speech

February 15, 1974
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Conference Paper
Author:
Clifford J. Weinstein
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Published in:
Proc. IEEE Symp. on Speech Recognition, 15-19 April 1974, pp. 54-67.
Topic:
speaker recognition
R&D area:
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Summary

A system for acoustic-phonetic analysis of continuous speech is being developed to serve as part of an automatic speech understanding system. The acoustic system accepts the speech waveform as an input and produces as output a string of phoneme-like units referred to as acoustic phonetic elements (APEL'S). This paper should be considered as a progress report, since the system is still under active development. The initial phase of the acoustic analysis consists of signal processing and parameter extraction, and includes spectrum analysis via linear prediction, computation of a number of parameters of the spectrum, and fundamental frequency extraction. This is followed by a preliminary segmentation of the speech into a few broad acoustic categories and formant tracking during vowel-like segments. The next phase consists of more detailed segmentation and classification intended to meet the needs of subsequent linguistic analysis. The preliminary segmentation and segment classification yield the following categories: vowel-like sound; volume dip within vowel-like sound; fricative-like sound; stop consonants, including silence or voice bar, and associated burst. These categories are produced by a deviation tree based upon energy measurements in selected frequency bands, derivatives and ratios of these measurements, a voicing detector, and a few editing rules. The more detailed classification algorithms include: 1) detection and identification of some diphthongs, semivowels, and nasals, through analysis of formant motions, positions, and amplitudes; 2) a vowel identifier, which determines three ranked choices for each vowel based on a comparison of the formant positions in the detected vowel segment to stored formant positions in a speaker-normalized vowel table; 3) a fricative identifier, which employs measurement of relative spectral energies in several bands to group the fricative segments into phoneme-like categories; 4) stop consonant classification based on the properties of the plosive burst. The above algorithms have been tested on a substantial corpus of continuous speech data. Performance results, as well as detailed descriptions of the algorithms are given.
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Summary

A system for acoustic-phonetic analysis of continuous speech is being developed to serve as part of an automatic speech understanding system. The acoustic system accepts the speech waveform as an input and produces as output a string of phoneme-like units referred to as acoustic phonetic elements (APEL'S). This paper should...

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A system for acoustic-phonetic analysis of continuous speech

Effects of finite register length in digital filtering and the fast Fourier transform

April 20, 1972
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Journal Article
Author:
Alan V. Oppenheim
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Published in:
Proceedings of the IEEE Vol. 60, No. 8, Aug 72, pp. 957-976.
Topic:
signal processing
R&D area:
R&D group:

Summary

When digital signal processing operations are implemented on a computer or with special-purpose hardware, errors and constraints due to finite word length are unavoidable. The main categories of finite register length effects are errors due to A/D conversion, errors due to roundoffs in the arithmetic, constraints on signal levels imposed by the need to prevent overflow, and quantization of system coefficients. The effects of finite register length on implementations of linear recursive difference equation digital filters, and the fast Fourier transform (FFT), are discussed in some detail. For these algorithms, the differing quantization effects of fixed point, floating point, and block floating point arithmetic are examined and compared. The paper is intended primarily as a tutorial review of a subject which has received considerable attention over the past few years. The groundwork is set through a discussion of the relationship between the binary representation of numbers and truncation or rounding, and a formulation of a statistical model for arithmetic roundoff. The analyses presented here are intended to illustrate techniques of working with particular models. Results of previous work are discussed and summarized when appropriate. Some examples are presented to indicate how the results developed for simple digital filters and the FFT can be applied to the analysis of more complicated systems which use these algorithms as building blocks.
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Summary

When digital signal processing operations are implemented on a computer or with special-purpose hardware, errors and constraints due to finite word length are unavoidable. The main categories of finite register length effects are errors due to A/D conversion, errors due to roundoffs in the arithmetic, constraints on signal levels imposed...

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Effects of finite register length in digital filtering and the fast Fourier transform
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