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Noise reduction based on spectral change

Published in:
Proc. of the 1997 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics, Session 8: Noise Reduction, 19-22 October 1997, 4 pages.

Summary

A noise reduction algorithm is designed for the aural enhancement of short-duration wideband signals. The signal of interest contains components possibly only a few milliseconds in duration and corrupted by nonstationary noise background. The essence of the enhancement technique is a Weiner filter that uses a desired signal spectrum whose estimation adapts to the "degree of stationarity" of the measured signal. The degree of stationarity is derived from a short-time spectral derivative measurement, motivated by sensitivity of biological systems to spectral change. Adaptive filter design tradeoffs are described, reflecting the accuracy of signal attack, background fidelity, and perceptual quality of the desired signal. Residual representations for binaural presentation are also considered.
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Summary

A noise reduction algorithm is designed for the aural enhancement of short-duration wideband signals. The signal of interest contains components possibly only a few milliseconds in duration and corrupted by nonstationary noise background. The essence of the enhancement technique is a Weiner filter that uses a desired signal spectrum whose...

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Embedded dual-rate sinusoidal transform coding

Published in:
Proc. IEEE Workshop on Speech Coding for Telecommunications Proc.: Back to Basics: Attacking Fundamental Problems in Speech Coding, 7-10 September 1997, pp. 33-34.

Summary

This paper describes the development of a dual-rate Sinusoidal Transformer Coder in which a 2400 b/s coder is embedded as a separate packet in the 4800 b/s bit stream. The underlying coding structure provides the flexibility necessary for multirate speech coding and multimedia applications.
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Summary

This paper describes the development of a dual-rate Sinusoidal Transformer Coder in which a 2400 b/s coder is embedded as a separate packet in the 4800 b/s bit stream. The underlying coding structure provides the flexibility necessary for multirate speech coding and multimedia applications.

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AM-FM separation using auditory-motivated filters

Published in:
IEEE Trans. Speech Audio Process., Vol. 5, No. 5, September 1997, pp. 465-480.

Summary

An approach to the joint estimation of sine-wave amplitude modulation (AM) and frequency modulation (FM) is described based on the transduction of frequency modulation into amplitude modulation by linear filters, being motivated by the hypothesis that the auditory system uses a similar transduction mechanism in measuring sine-wave FM. An AM-FM estimation is described that uses the amplitude envelope of the output of two transduction filters of piecewise-linear spectral shape. The piecewise-linear constraint is then relaxed, allowing a wider class of transduction-filter pairs for AM-FM separation under a monotonicity constraint of the filters' quotient. The particular case of Gaussian filters, and measured auditory filters, although not leading to a solution in closed form, provide for iterative AM-FM estimation. Solution stability analysis and error evaluation are performed and the FM transduction method is compared with the energy separation algorithm, based on the Teager energy operator, and the Hilbert transform method for AM-FM estimation. Finally, a generalization to two-dimensional (2-D) filters is described.
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Summary

An approach to the joint estimation of sine-wave amplitude modulation (AM) and frequency modulation (FM) is described based on the transduction of frequency modulation into amplitude modulation by linear filters, being motivated by the hypothesis that the auditory system uses a similar transduction mechanism in measuring sine-wave FM. An AM-FM...

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Fine structure features for speaker identification

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 2, Speech (Part II), 7-10 May 1996, pp. 689-692.

Summary

The performance of speaker identification (SID) systems can be improved by the addition of the rapidly varying "fine structure" features of formant amplitude and/or frequency modulation and multiple excitation pulses. This paper shows how the estimation of such fine structure features can be improved further by obtaining better estimates of formant frequency locations and uncovering various sources of error in the feature extraction systems. Most female telephone speech showed "spurious" formants, due to distortion in the telephone network. Nevertheless, SID performance was greatest with these spurious formants as formant estimates. A new feature has also been identified which can increase SID performance: cepstral coefficients from noise in the estimated excitation waveform. Finally, statistical tools have been developed to explore the relative importance of features used for SID, with the ultimate goal of uncovering the source of the features that provide SID performance improvement.
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Summary

The performance of speaker identification (SID) systems can be improved by the addition of the rapidly varying "fine structure" features of formant amplitude and/or frequency modulation and multiple excitation pulses. This paper shows how the estimation of such fine structure features can be improved further by obtaining better estimates of...

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Low rate coding of the spectral envelope using channel gains

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 2, 7-10 May 1996, pp. 769-772.

Summary

A dual rate embedded sinusoidal transform coder is described in which a core 14th order allpole coder operating at 2400 b/s is augmented with a set of channel gain residuals in order to operate at the higher 4800 b/s rate. The channel gains are a set of non-uniformly spaced samples of the spline envelope and constitute a lowpass estimate of the short-time vocal tract magnitude spectrum. The channel gain residuals represent the difference between the spline envelope and the quantized 14th order allpole spectrum at the channel gain frequencies. The channel gain residuals are coded using pitch dependent scalar quantization. Informal listening indicates that the quality of the embedded coder at 4800 b/s is comparable to that of an existing high quality 4800 b/s allpole coder.
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Summary

A dual rate embedded sinusoidal transform coder is described in which a core 14th order allpole coder operating at 2400 b/s is augmented with a set of channel gain residuals in order to operate at the higher 4800 b/s rate. The channel gains are a set of non-uniformly spaced samples...

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A subband approach to time-scale expansion of complex acoustic signals

Published in:
IEEE Trans. Speech Audio Process., Vol. 3, No. 6, November 1995, pp. 515-519.

Summary

A new approach to time-scale expansion of short-duration complex acoustic signals is introduced. Using a subband signal representation, channel phases are selected to preserve a desired time-scaled temporal envelope. The phase representation is derived from locations of events that occur within filter bank outputs. A frame-based generalization of the method imposes phase consistency across consecutive synthesis frames. The method is applied to synthetic and actual complex acoustic signals consisting of closely spaced rapidly damped sine wave. Time-frequency resolution limitations are discussed.
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Summary

A new approach to time-scale expansion of short-duration complex acoustic signals is introduced. Using a subband signal representation, channel phases are selected to preserve a desired time-scaled temporal envelope. The phase representation is derived from locations of events that occur within filter bank outputs. A frame-based generalization of the method...

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Time-scale modification with inconsistent constraints

Published in:
Proc. 1995 Workshop on Applications of Signal Processing to Audio Acoustics, 15-18 October 1995.

Summary

A set theoretic estimation approach is introduced for timescale modification of complex acoustic signals. The method determines a signal that meets, in a least-squared error sense, desired temporal and spectral envelope constraints that are inconsistent. These constraints are generalized within the set theoretic framework to include other signal characteristics such as instantaneous frequency and group delay. The approach can enhance acoustic signals consisting of closely-spaced sequential time components, and is applicable to biological, underwater, and music sound processing.
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Summary

A set theoretic estimation approach is introduced for timescale modification of complex acoustic signals. The method determines a signal that meets, in a least-squared error sense, desired temporal and spectral envelope constraints that are inconsistent. These constraints are generalized within the set theoretic framework to include other signal characteristics such...

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Sine-wave amplitude coding using a mixed LSF/PARCOR representation

Published in:
Proc. 1995 IEEE Workshop on Speech Coding for Telecommunications, 20-22 Spetember 1995, pp. 77-8.

Summary

An all-pole model of the speech spectral envelope is used to code the sine-wave amplitudes in the Sinusoidal Transform Coder. While line spectral frequencies (LSFs) are currently used to represent this all-pole model, it is shown that a mixture of line spectral frequencies and partial correlation (PARCOR) coefficients can be used to reduce complexity without a loss in quantization efficiency. Objective and subjective measures demonstrate that speech quality is maintained. In addition, the use of split vector quantization is shown to substantially reduce the number of bits needed to code the all-pole model.
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Summary

An all-pole model of the speech spectral envelope is used to code the sine-wave amplitudes in the Sinusoidal Transform Coder. While line spectral frequencies (LSFs) are currently used to represent this all-pole model, it is shown that a mixture of line spectral frequencies and partial correlation (PARCOR) coefficients can be...

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Measuring fine structure in speech: application to speaker identification

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, 9-12 May 1995, pp. 325-328.

Summary

The performance of systems for speaker identification (SID) can be quite good with clean speech, though much lower with degraded speech. Thus it is useful to search for new features for SID, particularly features that are robust over a degraded channel. This paper investigates features that are based on amplitude and frequency modulations of speech formants, high resolution measurement of fundamental frequency and location of "secondary pulses," measured using a high-resolution energy operator. When these features are added to traditional features using an existing SID system with a 168 speaker telephone speech database, SID performance improved by as much as 4% for male speakers and 8.2% for female speakers.
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Summary

The performance of systems for speaker identification (SID) can be quite good with clean speech, though much lower with degraded speech. Thus it is useful to search for new features for SID, particularly features that are robust over a degraded channel. This paper investigates features that are based on amplitude...

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The effects of telephone transmission degradations on speaker recognition performance

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, Speech, 9-12 May 1995, pp. 329-332.

Summary

The two largest factors affecting automatic speaker identification performance are the size of the population an the degradations introduced by noisy communication, channels (e.g., telephone transmission). To examine experimentally these two factors, this paper presents text-independent speaker identification results for varying speaker population sizes up to 630 speakers for both clean, wideband speech and telephone speech. A system based on Gaussian mixture speaker identification and experiments are conducted on the TIMIT and NTIMIT databases. This is believed to be the first speaker identification experiments on the complete 630 speaker TIMIT and NTIMIT databases and the largest text-independent speaker identification task reported to date. Identification accuracies of 99.5% and 60.7% are achieved on the TIMIT and NTIMIT databases, respectively. This paper also presents experiments which examine and attempt to quantify the performance loss associated with various telephone degradations by systematically degrading the TIMIT speech in a manner consistent with measured NTIMIT degradations and measuring the performance loss at each step. It is found that the standard degradations of filtering and additive noise do not account for all of the performance gap between the TIMIT and NTIMIT data. Measurements of nonlinear microphone distortions are also...
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Summary

The two largest factors affecting automatic speaker identification performance are the size of the population an the degradations introduced by noisy communication, channels (e.g., telephone transmission). To examine experimentally these two factors, this paper presents text-independent speaker identification results for varying speaker population sizes up to 630 speakers for both...

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