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Peak-to-rms reduction of speech based on a sinusoidal model

Published in:
IEEE Trans. Signal Process., Vol. 39, No. 2, February 1991, pp. 273-288.

Summary

In a number of applications, a speech waveform is processed using phase dispersion and amplitude compression to reduce its peak-to-rms ratio so as to increase loudness and intelligibility while minimizing perceived distortion. In this paper, a sinusoidal-based analysis/synthesis system is used to apply a radar design solution to the problem of dispersing the phase of a speech waveform. Unlike conventional methods of phase dispersion, this solution technique adapts dynamically to the pitch and spectral characteristics of the speech, while maintaining the original spectral envelope. The solution can also be used to drive the sine-wave amplitude modification for amplitude compression, and is coupled to the desired shaping of the speech spectrum. The new dispersion solution, when integrated with amplitude compression, results in a significant reduction in the peak-to-rms ratio of the speech waveform with acceptable loss in quality. Application of a real-time prototype sine-wave preprocessor to AM radio broadcasting is described.
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Summary

In a number of applications, a speech waveform is processed using phase dispersion and amplitude compression to reduce its peak-to-rms ratio so as to increase loudness and intelligibility while minimizing perceived distortion. In this paper, a sinusoidal-based analysis/synthesis system is used to apply a radar design solution to the problem...

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Short-time signal representation by nonlinear difference equations

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 3, Digital Signal Processing, 3-6 April 1990, pp. 1551-1554.

Summary

The solution of a nonlinear difference equation can take on complicated deterministic behavior which appears to be random for certain values of the equation's coefficients. Due to the sensitivities to initial conditions of the output of such "chaotic" systems, it is difficult to duplicate the waveform structure by parameter analysis and waveform synthesis techniques. In this paper, methods are investigated for short-time analysis and synthesis of signals from a class of second-order difference equations with a cubic nonlinearity. In analysis, two methods are explored for estimating equation coefficients: (1) prediction error minimization (a linear estimation problem) and (2) waveform error minimization (a nonlinear estimation problem). In the latter case, which improves on the prediction error solution, an iterative analysis-by-synthesis method is derived which allows as free variables initial conditions, as well as equation coefficients. Parameter estimates from these techniques are used in sequential short-time synthesis procedures. Possible application to modeling "quasi-periodic" behavior in speech waveforms is discussed.
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Summary

The solution of a nonlinear difference equation can take on complicated deterministic behavior which appears to be random for certain values of the equation's coefficients. Due to the sensitivities to initial conditions of the output of such "chaotic" systems, it is difficult to duplicate the waveform structure by parameter analysis...

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Noise reduction using a soft-decision sine-wave vector quantizer

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 2, Speech Processing 2; VLSI, Audio and Electroacoustics, 3-6 April 1990, pp. 821-824.

Summary

The need for noise reduction arises in speech communication channels, such as ground-to-air transmission and ground-based cellular radio, to improve vocoder quality and speech recognition accuracy. In this paper, noise reduction is performed in the context of a high-quality harmonic serc-phase sine-wave analysis/synthesis system which is characterized by sine-wave amplitudes, a voicing probability, and a fundamental frequency. Least-squared error estimation of a harmonic sine-wave representation leads to a "soft decision" template estimate consisting of sine-wave amplitudes and a voicing probability. The least-squares solution is modified to use template-matching with "nearest neighbors." The reconstruction is improved by using the modified least-squares solution only in spectral regions with low signal-to-noise ratio. The results, although preliminary, provide evidence that harmonic zero-phase sine-wave analysis/synthesis, combined with effective estimation of sine-wave amplitudes and probability of voicing, offers a promising approach to noise reduction.
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Summary

The need for noise reduction arises in speech communication channels, such as ground-to-air transmission and ground-based cellular radio, to improve vocoder quality and speech recognition accuracy. In this paper, noise reduction is performed in the context of a high-quality harmonic serc-phase sine-wave analysis/synthesis system which is characterized by sine-wave amplitudes...

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An approach to co-channel talker interference suppression using a sinusoidal model for speech

Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. 38, No. 1, January 1990, pp. 56-59.

Summary

This paper describes a new approach to co-channel talker interference suppression on a sinusoidal representation of speech. The technique fits a sinusoidal model to additive vocalic speech segments such that the least mean-squared error between the model and the summed waveforms is obtained. Enhancement is achieved by synthesizing a waveform from the sine waves attributed to the desired speaker. Least-squares estimation is applied to obtain sine-wave amplitudes and phases of both talkers, based on either a priori sine-wave frequencies or a priori fundamental frequency contours. When the frequencies of the two waveforms are closely spaced, the performance is significantly improved by exploiting the time evolution of the sinusoidal parameters across multiple analysis frames. The least-squared error approach is also extended, under restricted conditions, to estimate fundamental frequency contours of both speakers from the summed waveforms. The results obtained, although limited in their scope, provide evidence that the sinusoidal analysis/synthesis model with effective parameter estimation techniques offers a promising approach to the problem of co-channel talker interference suppression over a range of conditions.
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Summary

This paper describes a new approach to co-channel talker interference suppression on a sinusoidal representation of speech. The technique fits a sinusoidal model to additive vocalic speech segments such that the least mean-squared error between the model and the summed waveforms is obtained. Enhancement is achieved by synthesizing a waveform...

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Far-echo cancellation in the presence of frequency offset (full duplex modem)

Published in:
IEEE Trans. Commun., Vol. 37, No. 6, June 1989, pp. 635-644.

Summary

In this paper, we present a design for a full-duplex echo-cancelling data modem based on a combined adaptive reference algorithm and adaptive channel equalizer. The adaptive reference algorithm has the advantage that interference to the echo canceller caused by the far-end signal can be eliminated by subtracting an estimate of the far-end signal based on receiver decisions. This technique provides a new approach for full-duplex far-echo cancellation in which the far echo can be cancelled in spite of carrier frequency offset. To estimate the frequency offset, the system uses a separate receiver structure for the far echo which provides equalization of the far-echo channel and tracks the frequency offset in the far echo. The feasibility of the echo-cancelling algorithms is demonstrated by computer simulation with realistic channel distortions and with 4800 bits/s data transmission at which rate frequency offset in the far echo becomes important.
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Summary

In this paper, we present a design for a full-duplex echo-cancelling data modem based on a combined adaptive reference algorithm and adaptive channel equalizer. The adaptive reference algorithm has the advantage that interference to the echo canceller caused by the far-end signal can be eliminated by subtracting an estimate of...

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Phase coherence in speech reconstruction for enhancement and coding applications

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, Speech Processing 1, 23-26 May 1989, pp. 207-209.

Summary

It has been shown that an analysis-synthesis system based on a sinusoidal representation leads to synthetic speech that is essentially perceptually indistinguishable from the original. A change in speech quality has been observed, however, when the phase relation of the sine waves is altered. This occurs in practice when sine waves are processed for speech enhancement (e.g., time-scale modification and reducing peak-to-RMS ratio) and for speech coding. This paper describes a zero-phase sinusoidal analysis-synthesis system which generates natural-sounding speech without the requirement of vocal tract phase. The method provides a basis for improving sound quality by providing different levels of phase coherence in speech reconstruction for time-scale modification, for a baseline system for coding, and for reducing the peak-to-RMS ration by dispersion.
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Summary

It has been shown that an analysis-synthesis system based on a sinusoidal representation leads to synthetic speech that is essentially perceptually indistinguishable from the original. A change in speech quality has been observed, however, when the phase relation of the sine waves is altered. This occurs in practice when sine...

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Mixed-phase deconvolution of speech based on a sine-wave model

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 2, 6-9 April 1987, pp. 649-652.

Summary

This paper describes a new method of deconvolving the vocal cord excitation and vocal tract system response. The technique relies on a sine-wave representation of the speech waveform and forms the basis of an analysis-synthesis method which yields synthetic speech essentially indistinguishable from the original. Unlike an earlier sinusoidal analysis-synthesis technique that used a minimum-phase system estimate, the approach in this paper generates a "mixed-phase" system estimate and thus an improved decomposition of excitation and system components. Since a mixed-phase system estimate is removed from the speech waveform, the resulting excitation residual is less dispersed than the previous sinusoidal-based excitation estimate of the more commonly used linear prediction residual. A method of time-varying linear filtering is given as an alternative to sinusoidal reconstruction, similar to conventional time-domain synthesis used in certain vocoders, but without the requirement of pitch and voicing decisions. Finally, speech modification with a mixed-phase system estimate is shown to be capable of more closely preserving waveform shape in time-scale and pitch transformations than the earlier approach.
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Summary

This paper describes a new method of deconvolving the vocal cord excitation and vocal tract system response. The technique relies on a sine-wave representation of the speech waveform and forms the basis of an analysis-synthesis method which yields synthetic speech essentially indistinguishable from the original. Unlike an earlier sinusoidal analysis-synthesis...

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Speech transformations based on a sinusoidal representation

Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-34, No. 6, December 1986, pp. 1449-1464.

Summary

In this paper a new speech analysis/synthesis technique is presented which provides the basis for a general class of speech transformations including time-scale modification, frequency scaling, and pitch modification. These modifications can be performed with a time-varying change, permitting continuous adjustment of a speaker's fundamental frequency rate of articulation. The method is based on a sinusoidal representation of the speech production mechanism which has been shown to produce synthetic speech that preserves the waveform shape and is perceptually indistinguishable from the original. Although the analysis/synthesis system was originally designed for single speaker signals, it is also capable ot recovering and modifying non-speech signals such as music, multiple speakers, marine biologic sounds, and speakers in the presence of interferences such as noise and musical backgrounds.
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Summary

In this paper a new speech analysis/synthesis technique is presented which provides the basis for a general class of speech transformations including time-scale modification, frequency scaling, and pitch modification. These modifications can be performed with a time-varying change, permitting continuous adjustment of a speaker's fundamental frequency rate of articulation. The...

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Speech analysis/synthesis based on a sinusoidal representation

Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-34, No. 4, August 1986, pp. 744-754.

Summary

A sinusoidal model for the speech waveform is used to develop a new analysis/synthesis technique that is characterized by the amplitudes, frequencies, and phases of the component sine waves. These parameters are estimated from the short-time Fourier transform using a simple peak-picking algorithm. Rapid changes in the highly resolved spectral components are tracked using the concept of "birth" and "death" of the underlying sine waves. For a given frequency track a cubic function is used to unwrap and interpolate the phase such that the phase track is maximally smooth. This phase function is applied to a sine-wave generator, which is amplitude modulated and added to the other sine waves to give the final speech output. The resulting synthetic waveform preserves the general waveform shape and is essentially perceptually indistinguishable from the original speech. Furthermore, in the presence of noise the perceptual characteristics of the speech as well as the noise are maintained. In addition, it was found that the representation was sufficiently general that high-quality reproduction was obtained for a larger class of inputs including: two overpallping, superposed speech waveforms; music waveforms; speech in musical backgrounds; and certain marine biologic sounds. Finally, the analysis/synthesis system forms the basis for new approaches to the problems of speech transformations including time-scale and pitch-scale modification, and midrate speech coding.
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Summary

A sinusoidal model for the speech waveform is used to develop a new analysis/synthesis technique that is characterized by the amplitudes, frequencies, and phases of the component sine waves. These parameters are estimated from the short-time Fourier transform using a simple peak-picking algorithm. Rapid changes in the highly resolved spectral...

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Frequency sampling of the short-time Fourier-transform magnitude for signal reconstruction

Published in:
J. Opt. Soc. Amer., Vol. 73, November 1983, pp. 1523- 1526.

Summary

Unique recovery of a signal from the magnitude (modulus) of the Fourier transform has been of long-standing interest in image and optical processing in which Fourier-transform phase is lost or difficult to measure. We investigate an alternative problem of recovering a signal from the Fourier-transform magnitude of overlapping regions of the signal, i.e., from the short-time (or -space) Fourier-transform magnitude. Recently it was established that a discrete-time signal x (n) can be uniquely obtained under mild restrictions from its short-time Fourier-transform magnitude. In this paper we extend this result to the case when the short-time Fourier-transform magnitude is known at only one or two frequencies for each n. We also present a recursive algorithm for recovering a sequence from such samples and demonstrate the algorithm with an example.
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Summary

Unique recovery of a signal from the magnitude (modulus) of the Fourier transform has been of long-standing interest in image and optical processing in which Fourier-transform phase is lost or difficult to measure. We investigate an alternative problem of recovering a signal from the Fourier-transform magnitude of overlapping regions of...

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