Publications

Refine Results

(Filters Applied) Clear All

A comparison of signal processing front ends for automatic word recognition

Published in:
IEEE Trans. Speech Audio Process., Vol. 3, No. 4, July 1995, pp. 286-293.

Summary

This paper compares the word error rate of a speech recognizer using several signal processing front ends based on auditory properties. Front ends were compared with a control mel filter banks (MFB) based cepstral front end in clean speech and with speech degraded by noise and spectral variability, using the TI-105 isolated word database. MFB recognition error rates ranged from 0.5 to 3.1%,, and the reduction in error rates provided by auditory models was less than 0.5 percentage points. Some earlier studies that demonstrated considerably more improvement with auditory models used linear predictive coding (LPC) based control front ends. This paper shows that MFB cepstra significantly outperform LPC cepstra under noisy conditions. Techniques using an optimal linear combination of features for data reduction were also evaluated.
READ LESS

Summary

This paper compares the word error rate of a speech recognizer using several signal processing front ends based on auditory properties. Front ends were compared with a control mel filter banks (MFB) based cepstral front end in clean speech and with speech degraded by noise and spectral variability, using the...

READ MORE

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.
READ LESS

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...

READ MORE

Language identification using phoneme recognition and phonotactic language modeling

Author:
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, Vol. 5, ICASSP, 9-12 May 1995, pp. 3503-3506.

Summary

A language identification technique using multiple single-language phoneme recognizers followed by n-gram language models yielded to performance at the March 1994 NIST language identification evaluation. Since the NIST evaluation, work has been aimed at further improving performance by using the acoustic likelihoods emitted from gender-dependent phoneme recognizers to weight the phonotactic likelihoods output from gender-dependent language models. We have investigated the effect of restricting processing to the most highly discriminating n-grams, and we have also added explicit duration modeling at the phonotactic level. On the OGI Multi-language Telephone Speech Corpus, accuracy on an 11-language identification task has risen to 89% on 45-s utterances and 79% on 10-s utterances. Two-language classification accuracy is 98% and 95% for the 45-s and 10-s utterance, respectively. Finally, we have started to apply these same techniques to the problem of dialect identification.
READ LESS

Summary

A language identification technique using multiple single-language phoneme recognizers followed by n-gram language models yielded to performance at the March 1994 NIST language identification evaluation. Since the NIST evaluation, work has been aimed at further improving performance by using the acoustic likelihoods emitted from gender-dependent phoneme recognizers to weight the...

READ MORE

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...
READ LESS

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...

READ MORE

Large population speaker identification using clean and telephone speech

Published in:
IEEE Signal Process. Lett., Vol. 2, No. 3, March 1995, pp. 46-48.

Summary

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 models is used for speaker identification, and experiments are conducted on the TIMIT and NTIMIT databases. The TIMIT results show large population performance under near-ideal conditions, and the NTIMIT results show the corresponding accuracy loss due to telephone transmission. These are 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% were achieved on the TIMIT and NTIMIT databases, respectively.
READ LESS

Summary

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 models is used for speaker identification, and experiments are conducted on the TIMIT and NTIMIT databases. The TIMIT results...

READ MORE

Robust text-independent speaker identification using Gaussian mixture speaker models

Published in:
IEEE Trans. Speech Audio Process., Vol. 3, No. 1, January 1995, pp. 72-83.

Summary

This paper introduces and motivates the use of Gaussian mixture models (GMM) for robust text-independent speaker identification. The individual Gaussian components of a GMM are shown to represent some general speaker-dependent spectral shapes that are effective for modeling speaker identify. The focus of this work is on applications which require high identification rates using short utterance from unconstrained conversational speech and robustness to degradations produced by transmission over a telephone channel. A complete experimental evaluation of the Gaussian mixture speaker model is conducted on a 49 speaker, conversational telephone speech database. The experiments examine algorithmic issues (initializations, variance limiting, model order selection), spectral variability robustness techniques, large population performance, and comparisons to other speaker modeling techniques (uni-modal Gaussian, VQ codebook, tied Gaussian mixture, and radial basis functions). The Gaussian mixture speaker model attains 96.8% identification accuracy using 5 second clean speech utterances and 80.8% accuracy using 15 second telephone speech utterances with a 49 speaker population and is shown to outperform the other speaker modeling techniques on an identical 16 speaker telephone speech task.
READ LESS

Summary

This paper introduces and motivates the use of Gaussian mixture models (GMM) for robust text-independent speaker identification. The individual Gaussian components of a GMM are shown to represent some general speaker-dependent spectral shapes that are effective for modeling speaker identify. The focus of this work is on applications which require...

READ MORE

Sinusoidal coding

Published in:
Chapter 4 in Speech Coding and Synthesis, Elsevier Science Publishers, 1995, pp. 121-173.

Summary

This chapter summarizes the sinewave-based pitch extractor, and the high-order all-pole modelling techniques that provided the basis for the multirate Sinusoidal Transform Coder and its application to multi-speaker conferencing.
READ LESS

Summary

This chapter summarizes the sinewave-based pitch extractor, and the high-order all-pole modelling techniques that provided the basis for the multirate Sinusoidal Transform Coder and its application to multi-speaker conferencing.

READ MORE

Speaker identification and verification using Gaussian mixture speaker models

Published in:
Speech Commun., Vol. 17, 1995, pp. 91-108.

Summary

This paper presents high performance speaker identification and verification systems based on Gaussian mixture speaker models: robust, statistically based representations of speaker identification. The identification system is a maximum likelihood classifier and the verification system is a likelihood ratio hypothesis tester using background speaker normalization. The systems are evaluated on four publically available speech databases: TIMIT, NTIMIT, Switchboard and YOHO. The different levels of degradation and variabilities found in these databases allow the examination of system performance for different task domains. Constraints on the speech range from vocabulary-dependent to extemporaneous and speech quality varies from near-ideal, clean speech to noisy, telephone speech. Closed set identification accuracies on the 630 speaker TIMIT and NTIMIT databases were 99.5% and 60.7% respectively. On a 113 speaker population from the Switchboard database the identification accuracy was 82.8%. Global threshold equal error rates of 0.24%, 7.19%, 5.15% and 0.51% were obtained in verification experiments on the TIMIT, NTIMIT, Switchboard and YOHO databases, respectively.
READ LESS

Summary

This paper presents high performance speaker identification and verification systems based on Gaussian mixture speaker models: robust, statistically based representations of speaker identification. The identification system is a maximum likelihood classifier and the verification system is a likelihood ratio hypothesis tester using background speaker normalization. The systems are evaluated on...

READ MORE

Energy onset times for speaker identification

Published in:
IEEE Signal Process. Lett., Vol. 1, No. 11, November 1994, pp. 160-162.

Summary

Onset times of resonant energy pulses are measured with the high-resolution Teager operator and used as features in the Reynolds Gaussian-mixture speaker identification algorithm. Feature sets are constructed with primary pitch and secondary pulse locations derived from low and high speech formants. Preliminary testing was performed with a confusable 40-speaker subset from the NTIMIT (telephone channel) database. Speaker identification improved from 55 to 70% correct classification when the full set of new resonant energy-based features were added as an independent stream to conventional mel-cepstra.
READ LESS

Summary

Onset times of resonant energy pulses are measured with the high-resolution Teager operator and used as features in the Reynolds Gaussian-mixture speaker identification algorithm. Feature sets are constructed with primary pitch and secondary pulse locations derived from low and high speech formants. Preliminary testing was performed with a confusable 40-speaker...

READ MORE

Formant AM-FM for speaker identification

Published in:
Proc. IEEE-SP Int. Symp. on Time-Frequency and Time-Scale Analysis, 25-28 October 1994, pp. 608-611.

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 robust over a degraded channel. This paper investigates features that are based on amplitude and frequency modulations of speech formants. Such modulations are measured using a high-resolution energy operator and related algorithms for recovering amplitude and frequency from an AM-FM signal. When these features are added to traditional features using an existing SID system with a telephone speech database, SID performance improved by as much as 15%. Energy onset time measurements that yielded improved SID performance are also discussed.
READ LESS

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 robust over a...

READ MORE