NASA Scientific and Technical Aerospace Reports

In AM radio it is of interest to increase the perceived loudness of transmitted speech in order to increase the effective range
of the radio broadcasts. In particular, shortwave radio broadcasts are subject to a variety of natural and men-made interference
and noise which can severely limit the signal-to-noise ratio of the received signal. Often, broadcasting is done using very
powerful and expensive transmitters and it is of interest to determine whether or not signal processing techniques could
provide a more cost-effective increase of the broadcast range. In addition to developing digital signal processing (DSP)
algorithms [1,2] for enhancing the perceived loudness of speech, we are developing a facility to evaluate the effectiveness of
the audio preprocessing algorithms. ]his evaluation will eventually include subjective measures of processed speech quality,
quantitative measures of speech intelligibility, and measurement of the peak/rms ratio of the processed speech. ]his paper
describes a system that has been developed for automatically measuring the enhancement in the peak/rms ratio that a given
audio processor provides, and gives some preliminary results on the performance of three commercial audio processors and
two rather simple processors. An important outcome of the results is the establishment of an engineering goal for an improved
digital signal processing (DSP) speech enhancement algorithm.
Author
Signal Processing; Signal to Noise Ratios; Broadcasting; Dynamic Range
20060001697 Zoran Corp., Santa Clara, CA, USA
A Novel VLSI Digital Signal Processor Architecture for High-Speed Vector and Transform Operations
Taylor, David M.; Retter, Rafi; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87);
Volume 1; 1987, pp. 13.5.1 - 13.5.4; In English; See also 20060001583; Copyright; Avail.: Other Sources
This paper presents a new concept in monolithic digital signal processing integrated circuits. The device is called a Vector
Signal Processor (VSP) and is manufactured by Zoran Corporation. It achieves extremely high signal processing throughput
in both the time and frequency domains for many standard DSP operations. It introduces a number of architectural features
unique to digital signal processing components. Among these are: a ‘high-level’ instruction set which allows programming at
the functional level, embedded hardware to support block floating-point arithmetic, and the ability to easily parallel multiple
devices to achieve signal processing throughput even higher than that attainable with a single device. The combination of these
features and a powerful applications development environment make the VSP easy to both use and program for developing
sophisticated, high-performance applications.
Author
Signal Processing; Very Large Scale Integration; Architecture (Computers)
20060001699 Georgia Inst. of Tech., Atlanta, GA, USA
Iterative Speech Enhancement with Spectral Constraints
Hansen, John H.; Clements, Mark A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP
‘87); Volume 1; 1987, pp. 6.7.1 - 6.7.4; In English; See also 20060001583; Copyright; Avail.: Other Sources
A new and improved iterative speech enhancement technique based on spectral constraints is presented in this paper. The
iterative technique, originally formulated by Lim and Oppenheim, attempts to solve for the maximum likelihood estimate of
a speech waveform in additive white noise. The new approach applies inter- and intra-frame spectral constraints to ensure
convergence to reasonable values and hence improve speech quality. An extremely efficient technique for applying these
constraints is in the use of line spectral pair (LSP) coefficients. The inter-frame constraints ensures more speech-like formant
trajectories than those found in the unconstrained approach. Results from speech degraded by additive white Gaussian noise
show noticeable quality improvement.
Author
Speech Recognition; Maximum Likelihood Estimates; Convergence; Spectra; Augmentation
20060001700 GTE Labs., Inc., Waltham, MA, USA
Enhancement of Block-Coded Speech
Veeneman, D. E.; Mazor, B.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87);
Volume 1; 1987, pp. 6.9.1 - 6.9.4; In English; See also 20060001583; Copyright; Avail.: Other Sources
This paper describes a very effective method for reducing framing noise associated with block-by-block speech coding
algorithms. It is a post-coding speech enhancement procedure designed specifically to combat this type of noise. The
procedure is based on a fully adaptive comb filter which is adapted in synchrony with the coding algorithm, s frame boundary.
An optimal (minimum mean-squared-error sense) linear prediction approach is applied to determine the filter coefficients and
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