Voiced speech preprocessing employing waveform interpolation...

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

Reexamination Certificate

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Details

C704S203000, C704S233000, C704S208000, C704S275000

Reexamination Certificate

active

06738739

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to speech coding, and more particularly, to a system that performs speech pre-processing.
2. Related Art
Speech coding systems often do not operate at low bandwidths. When the bandwidth of a speech coding system is reduced, the perceptual quality of its output, a synthesized speech, is often reduced. In spite of this loss, there is an effort to reduce speech coding bandwidths.
Some speech coding systems perform strict waveform matching using code excited linear prediction (CELP) at low bandwidths such as 4 kbit/s. The waveform matching used by these systems do not always accurately encode and decode speech signals due to the system's limited capacity. This invention provides an efficient speech coding system and a method that modifies an original speech signal in transition areas, and accurately encodes and decodes the modified speech signal to keep the perceptually important features of a speech signal.
SUMMARY
A speech codec includes a classifier and a periodic smoothing circuit. The classifier processes a transition region that separates portions of a speech signal. The periodic smoothing circuit uses at least an interpolated pitch lag and/or a constant pitch lag to smooth the transition region that is represented by a residual signal, a weighted signal, or a portion of an unconditioned speech signal. The pitch track corresponds to the voiced portion of the speech signal.
In one aspect, the periodic smoothing circuit selects either a forward pitch extension or a backward pitch extension to smooth the transition region between two periodic signals. The transition region can extend through multiple frames and may include an unvoiced portion. The periodic smoothing circuit smoothes the transition region between these signals in the time domain using a waveform interpolation circuit, or in the frequency domain using a harmonic circuit. The smoothing may occur when a long term pre-processing circuit or a long term processing circuit fails or when an irregular voiced speech portion is detected.
In another aspect, the periodic smoothing circuit smoothes the transition region between a periodic portion of a speech signal and other portions of that signal. In this aspect, smoothing occurs in the time domain using the waveform interpolation circuit or in the frequency domain using the harmonic circuit. The classifier uses a pitch lag, a linear prediction coefficient, an energy level, a normalized pitch correlation, and/or other parameters to classify the speech signal.
Other systems, methods, features and advantages of the invention will become apparent to one with skill in the art upon examination of the following figures and detailed description. It is intended that all such additional systems, methods, features and advantages be included within this description, be within the scope of the invention, and be protected by the accompanying claims.


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Marques et al (“Harmonic Coding at 4.8 kb/s”, International Conference on Acoustics, Speech, and Signal Processing, Apr. 1990) harmonically related frequency use extend to unvoiced and transition regions for large frame length.*
Jiang et al (“Kbps-2.4 Kbps Low Complexity Interpolative Vocoder”, International Conference on Communication Technology Oct. 1998) interpolative speech coding algorithm including one-frame look ahead pitch smoothing.*
Kleijn et al (“A Low-Complexity Waveform Interpolation Coder”, IEEE International Conference on Acoustics, Speech, and Signal Processing, May 1996) addresses waveform smoothing.*
Burnett I S et al: “A Mixed Prototype Waveform/ CELP Coder for Sub 3 kbit/s” Statistical Signal and Array Processing, Minneapolis, Apr. 27-30, 1993, Proceedings of the International Conference on Acoustics, Speech, and Signal Processing (ICASSP), New York, IEEE, US, vol. 4, Apr. 27, 1993, pp. 175-178, XP010110423, ISBN: 0-7803-0946-4, chapters 2, 2.1—2.3, chapter 5, lines 1-7.

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