Publications

Refine Results

(Filters Applied) Clear All

Tagged As

An overview of automatic speaker recognition technology

May 13, 2002
  |
Conference Paper
Author:
Douglas A. Reynolds
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. IV, 13-17 May 2002, pp. IV-4072 - IV-4075.
Topic:
speech processing
R&D area:
R&D group:

Summary

In this paper we provide a brief overview of the area of speaker recognition, describing applications, underlying techniques and some indications, of performance. Following this overview we will discuss some of the strengths and weaknesses of current speaker recognition technologies and outline some potential future trends in research, development and applications conducting other speech interactions (background verification). As speaker and speech recognition system merge and speech recognition accuracy improves, the distinction between text- independent and -dependent applications will decrease. Of the two basic tasks, text-dependent speaker verification is currently
READ LESS

Summary

In this paper we provide a brief overview of the area of speaker recognition, describing applications, underlying techniques and some indications, of performance. Following this overview we will discuss some of the strengths and weaknesses of current speaker recognition technologies and outline some potential future trends in research, development and...

READ MORE
An overview of automatic speaker recognition technology

Language identification using Gaussian mixture model tokenization

January 1, 2002
  |
Conference Paper
Author:
Pedro A. Torres-Carrasquillo
…
Published in:
Proc. IEEE Int. Conf., on Acoustics, Speech and Signal Processing, ICASSP, Vol. I, 13-17 May 2002, pp. I-757 - I-760.
Topic:
language recognition
R&D area:
R&D group:

Summary

Phone tokenization followed by n-gram language modeling has consistently provided good results for the task of language identification. In this paper, this technique is generalized by using Gaussian mixture models as the basis for tokenizing. Performance results are presented for a system employing a GMM tokenizer in conjunction with multiple language processing and score combination techniques. On the 1996 CallFriend LID evaluation set, a 12-way closed set error rate of 17% was obtained.
READ LESS

Summary

Phone tokenization followed by n-gram language modeling has consistently provided good results for the task of language identification. In this paper, this technique is generalized by using Gaussian mixture models as the basis for tokenizing. Performance results are presented for a system employing a GMM tokenizer in conjunction with multiple...

READ MORE
Language identification using Gaussian mixture model tokenization

Speaker recognition from coded speech and the effects of score normalization

November 4, 2001
  |
Conference Paper
Author:
Robert B. Dunn
…
Published in:
Proc. Thirty-Fifth Asilomar Conf. on Signals, Systems and Computers, Vol. 2, 4-7 November 2001, pp. 1562-1567.
Topic:
speaker recognition
R&D area:
R&D group:

Summary

We investigate the effect of speech coding on automatic speaker recognition when training and testing conditions are matched and mismatched. Experiments used standard speech coding algorithms (GSM, G.729, G.723, MELP) and a speaker recognition system based on Gaussian mixture models adapted from a universal background model. There is little loss in recognition performance for toll quality speech coders and slightly more loss when lower quality speech coders are used. Speaker recognition from coded speech using handset dependent score normalization and test score normalization are examined. Both types of score normalization significantly improve performance, and can eliminate the performance loss that occurs when there is a mismatch between training and testing conditions.
READ LESS

Summary

We investigate the effect of speech coding on automatic speaker recognition when training and testing conditions are matched and mismatched. Experiments used standard speech coding algorithms (GSM, G.729, G.723, MELP) and a speaker recognition system based on Gaussian mixture models adapted from a universal background model. There is little loss...

READ MORE
Speaker recognition from coded speech and the effects of score normalization

Speaker recognition from coded speech in matched and mismatched conditions

June 18, 2001
  |
Conference Paper
Author:
Robert B. Dunn
…
Published in:
Proc. 2001: A Speaker Odyssey, The Speaker Recognition Workshop, 18-22 June 2001, pp. 115-20.
Topic:
speaker recognition
R&D area:
R&D group:

Summary

We investigate the effect of speech coding on automatic speaker recognition when training and testing conditions are matched and mismatched. Experiments use standard speech coding algorithms (GSM, G.729, G.723, MELP) and a speaker recognition system based on Gaussian mixture models adapted from a universal background model. There is little loss in recognition performance for toll quality speech coders and slightly more loss when lower quality speech coders are used. Speaker recognition from coded speech using handset dependent score normalization is examined, and we find that this significantly improves performance, particularly when there is a mismatch between training and testing conditions.
READ LESS

Summary

We investigate the effect of speech coding on automatic speaker recognition when training and testing conditions are matched and mismatched. Experiments use standard speech coding algorithms (GSM, G.729, G.723, MELP) and a speaker recognition system based on Gaussian mixture models adapted from a universal background model. There is little loss...

READ MORE
Speaker recognition from coded speech in matched and mismatched conditions

Speaker recognition using G.729 speech codec parameters

June 5, 2000
  |
Conference Paper
Author:
Thomas F. Quatieri
…
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. II, 5-9 June 2000, pp. 1089-1092.
Topic:
speaker recognition
R&D area:
R&D group:

Summary

Experiments in Gaussian-mixture-model speaker recognition from mel-filter bank energies (MFBs) of the G.729 codec all-pole spectral envelope, showed significant performance loss relative to the standard mel-cepstral coefficients of G.729 synthesized (coded) speech. In this paper, we investigate two approaches to recover speaker recognition performance from G.729 parameters, rather than deriving cepstra from MFBs of an all-pole spectrum. Specifically, the G.729 LSFs are converted to "direct" cepstral coefficients for which there exists a one-to-one correspondence with the LSFs. The G.729 residual is also considered; in particular, appending G.729 pitch as a single parameter to the direct cepstral coefficients gives further performance gain. The second nonparametric approach uses the original MFB paradigm, but adds harmonic striations to the G.729 all-pole spectral envelope. Although obtaining considerable performance gains with these methods, we have yet to match the performance of G.729 synthesized speech, motivating the need for representing additional fine structure of the G.729 residual.
READ LESS

Summary

Experiments in Gaussian-mixture-model speaker recognition from mel-filter bank energies (MFBs) of the G.729 codec all-pole spectral envelope, showed significant performance loss relative to the standard mel-cepstral coefficients of G.729 synthesized (coded) speech. In this paper, we investigate two approaches to recover speaker recognition performance from G.729 parameters, rather than deriving...

READ MORE
Speaker recognition using G.729 speech codec parameters

Speaker and language recognition using speech codec parameters

September 5, 1999
  |
Conference Paper
Author:
Thomas F. Quatieri
…
Published in:
EUROSPEECH 99, 5-10 September 1999.
Topic:
speaker recognition
R&D area:
R&D group:

Summary

In this paper, we investigate the effect of speech coding on speaker and language recognition tasks. Three coders were selected to cover a wide range of quality and bit rates: GSM at 12.2 kb/s, G.729 at 8 kb/s, and G.723.1 at 5.3 kb/s. Our objective is to measure recognition performance from either the synthesized speech or directly from the coder parameters themselves. We show that using speech synthesized from the three codecs, GMM-based speaker verification and phone-based language recognition performance generally degrades with coder bit rate, i.e., from GSM to G.729 to G.723.1, relative to an uncoded baseline. In addition, speaker verification for all codecs shows a performance decrease as the degree of mismatch between training and testing conditions increases, while language recognition exhibited no decrease in performance. We also present initial results in determining the relative importance of codec system components in their direct use for recognition tasks. For the G.729 codec, it is shown that removal of the post- filter in the decoder helps speaker verification performance under the mismatched condition. On the other hand, with use of G.729 LSF-based mel-cepstra, performance decreases under all conditions, indicating the need for a residual contribution to the feature representation.
READ LESS

Summary

In this paper, we investigate the effect of speech coding on speaker and language recognition tasks. Three coders were selected to cover a wide range of quality and bit rates: GSM at 12.2 kb/s, G.729 at 8 kb/s, and G.723.1 at 5.3 kb/s. Our objective is to measure recognition performance...

READ MORE
Speaker and language recognition using speech codec parameters

Embedded dual-rate sinusoidal transform coding

September 10, 1997
  |
Conference Paper
Author:
Elliot Singer
…
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.
Topic:
speech processing
R&D area:
R&D group:

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.
READ LESS

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.

READ MORE
Embedded dual-rate sinusoidal transform coding

Low rate coding of the spectral envelope using channel gains

May 7, 1996
  |
Conference Paper
Author:
Elliot Singer
…
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 2, 7-10 May 1996, pp. 769-772.
Topic:
speech processing
R&D area:
R&D group:

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.
READ LESS

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...

READ MORE
Low rate coding of the spectral envelope using channel gains

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

September 20, 1995
  |
Conference Paper
Author:
Robert B. Dunn
…
Published in:
Proc. 1995 IEEE Workshop on Speech Coding for Telecommunications, 20-22 Spetember 1995, pp. 77-8.
Topic:
speech processing
R&D area:
R&D group:

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.
READ LESS

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...

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

Sinusoidal coding

January 1, 1995
  |
Book Chapter
Author:
Robert J. McAulay
…
Published in:
Chapter 4 in Speech Coding and Synthesis, Elsevier Science Publishers, 1995, pp. 121-173.
Topic:
speech processing
R&D area:
R&D group:

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
Sinusoidal coding
21-30 of 34