Publications

Refine Results

(Filters Applied) Clear All

Adaptive short-time analysis-synthesis for speech enhancement

Published in:
2008 IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, 31 March - 4 April 2008.

Summary

In this paper we propose a multiresolution short-time analysis method for speech enhancement. It is well known that fixed resolution methods such as the traditional short-time Fourier transform do not generally match the time-frequency structure of the signal being analyzed resulting in poor estimates of the speech and noise spectra required for enhancement. This can lead to the reduction of quality in the enhanced signal through the introduction of artifacts such as musical noise. To counter these limitations, we propose an adaptive short-time analysis-synthesis scheme for speech enhancement in which the adaptation is based on a measure of local time-frequency concentration. Synthesis is made possible through a modified overlap-add procedure. Empirical results using voiced speech indicate a clear improvement over a fixed time-frequency resolution enhancement scheme both in terms of mean-square error and as indicated by informal listening tests.
READ LESS

Summary

In this paper we propose a multiresolution short-time analysis method for speech enhancement. It is well known that fixed resolution methods such as the traditional short-time Fourier transform do not generally match the time-frequency structure of the signal being analyzed resulting in poor estimates of the speech and noise spectra...

READ MORE

A covariance kernel for SVM language recognition

Published in:
ICASSP 2008, Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, 31 March - 4 April 2008, pp. 4141-4144.

Summary

Discriminative training for language recognition has been a key tool for improving system performance. In addition, recognition directly from shifted-delta cepstral features has proven effective. A recent successful example of this paradigm is SVM-based discrimination of languages based on GMM mean supervectors (GSVs). GSVs are created through MAP adaptation of a universal background model (UBM) GMM. This work proposes a novel extension to this idea by extending the supervector framework to the covariances of the UBM. We demonstrate a new SVM kernel including this covariance structure. In addition, we propose a method for pushing SVM model parameters back to GMM models. These GMM models can be used as an alternate form of scoring. The new approach is demonstrated on a fourteen language task with substantial performance improvements over prior techniques.
READ LESS

Summary

Discriminative training for language recognition has been a key tool for improving system performance. In addition, recognition directly from shifted-delta cepstral features has proven effective. A recent successful example of this paradigm is SVM-based discrimination of languages based on GMM mean supervectors (GSVs). GSVs are created through MAP adaptation of...

READ MORE

A multi-class MLLR kernel for SVM speaker recognition

Published in:
Proc. IEEE Int. Connf. on Acoustics, Speech and Signal Processing, ICASSP, 31 March - 4 April 2008, pp. 4117-4120.

Summary

Speaker recognition using support vector machines (SVMs) with features derived from generative models has been shown to perform well. Typically, a universal background model (UBM) is adapted to each utterance yielding a set of features that are used in an SVM. We consider the case where the UBM is a Gaussian mixture model (GMM), and maximum likelihood linear regression (MLLR) adaptation is used to adapt the means of the UBM. Recent work has examined this setup for the case where a global MLLR transform is applied to all the mixture components of the GMM UBM. This work produced positive results that warrant examining this setup with multi-class MLLR adaptation, which groups the UBM mixture components into classes and applies a different transform to each class. This paper extends the MLLR/GMM framework to the multiclass case. Experiments on the NIST SRE 2006 corpus show that multi-class MLLR improves on global MLLR and that the proposed system?s performance is comparable with state of the art systems.
READ LESS

Summary

Speaker recognition using support vector machines (SVMs) with features derived from generative models has been shown to perform well. Typically, a universal background model (UBM) is adapted to each utterance yielding a set of features that are used in an SVM. We consider the case where the UBM is a...

READ MORE

Exploiting temporal change in pitch in formant estimation

Published in:
Proc. IEEE Int. Conf. on Acoustic, Speech, and Signal Processes, ICASSP, 31 March - 4 April 2008, pp. 3929-3932.

Summary

This paper considers the problem of obtaining an accurate spectral representation of speech formant structure when the voicing source exhibits a high fundamental frequency. Our work is inspired by auditory perception and physiological modeling studies implicating the use of temporal changes in speech by humans. Specifically, we develop and assess signal processing schemes aimed at exploiting temporal change of pitch as a basis for formant estimation. Our methods are cast in a generalized framework of two-dimensional processing of speech and show quantitative improvements under certain conditions over representations derived from traditional and homomorphic linear prediction. We conclude by highlighting potential benefits of our framework in the particular application of speaker recognition with preliminary results indicating a performance gender-gap closure on subsets of the TIMIT corpus.
READ LESS

Summary

This paper considers the problem of obtaining an accurate spectral representation of speech formant structure when the voicing source exhibits a high fundamental frequency. Our work is inspired by auditory perception and physiological modeling studies implicating the use of temporal changes in speech by humans. Specifically, we develop and assess...

READ MORE

Language recognition with discriminative keyword selection

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, 31 March - 4 April 2008, pp. 4145-4148.

Summary

One commonly used approach for language recognition is to convert the input speech into a sequence of tokens such as words or phones and then to use these token sequences to determine the target language. The language classification is typically performed by extracting N-gram statistics from the token sequences and then using an N-gram language model or support vector machine (SVM) to perform the classification. One problem with these approaches is that the number of N-grams grows exponentially as the order N is increased. This is especially problematic for an SVM classifier as each utterance is represented as a distinct N-gram vector. In this paper we propose a novel approach for modeling higher order Ngrams using an SVM via an alternating filter-wrapper feature selection method. We demonstrate the effectiveness of this technique on the NIST 2007 language recognition task.
READ LESS

Summary

One commonly used approach for language recognition is to convert the input speech into a sequence of tokens such as words or phones and then to use these token sequences to determine the target language. The language classification is typically performed by extracting N-gram statistics from the token sequences and...

READ MORE

Multisensor very low bit rate speech coding using segment quantization

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, 31 March - 4 April 2008, pp. 3997-4000.

Summary

We present two approaches to noise robust very low bit rate speech coding using wideband MELP analysis/synthesis. Both methods exploit multiple acoustic and non-acoustic input sensors, using our previously-presented dynamic waveform fusion algorithm to simultaneously perform waveform fusion, noise suppression, and crosschannel noise cancellation. One coder uses a 600 bps scalable phonetic vocoder, with a phonetic speech recognizer followed by joint predictive vector quantization of the error in wideband MELP parameters. The second coder operates at 300 bps with fixed 80 ms segments, using novel variable-rate multistage matrix quantization techniques. Formal test results show that both coders achieve equivalent intelligibility to the 2.4 kbps NATO standard MELPe coder in harsh acoustic noise environments, at much lower bit rates, with only modest quality loss.
READ LESS

Summary

We present two approaches to noise robust very low bit rate speech coding using wideband MELP analysis/synthesis. Both methods exploit multiple acoustic and non-acoustic input sensors, using our previously-presented dynamic waveform fusion algorithm to simultaneously perform waveform fusion, noise suppression, and crosschannel noise cancellation. One coder uses a 600 bps...

READ MORE

Improved GMM-based language recognition using constrained MLLR transforms

Author:
Published in:
Proc. 33rd IEEE Int. Conf. on Acoustics, Speech, and SIgnal Processing, ICASSP, 30 March - 4 April 2008, pp. 4149-4152.

Summary

In this paper we describe the application of a feature-space transform based on constrained maximum likelihood linear regression for unsupervised compensation of channel and speaker variability to the language recognition problem. We show that use of such transforms can improve baseline GMM-based language recognition performance on the 2005 NIST Language Recognition Evaluation (LRE05) task by 38%. Furthermore, gains from CMLLR are additive with other modeling enhancements such as vocal tract length normalization (VTLN). Further improvement is obtained using discriminative training, and it is shown that a system using only CMLLR adaption produces state-of-the-art accuracy with decreased test-time computational cost than systems using VTLN.
READ LESS

Summary

In this paper we describe the application of a feature-space transform based on constrained maximum likelihood linear regression for unsupervised compensation of channel and speaker variability to the language recognition problem. We show that use of such transforms can improve baseline GMM-based language recognition performance on the 2005 NIST Language...

READ MORE

Spectral representations of nonmodal phonation

Published in:
IEEE Trans. Audio, Speech, Language Proc., Vol. 16, No. 1, January 2008, pp. 34-46.

Summary

Regions of nonmodal phonation, which exhibit deviations from uniform glottal-pulse periods and amplitudes, occur often in speech and convey information about linguistic content, speaker identity, and vocal health. Some aspects of these deviations are random, including small perturbations, known as jitter and shimmer, as well as more significant aperiodicities. Other aspects are deterministic, including repeating patterns of fluctuations such as diplophonia and triplophonia. These deviations are often the source of misinterpretation of the spectrum. In this paper, we introduce a general signal-processing framework for interpreting the effects of both stochastic and deterministic aspects of nonmodality on the short-time spectrum. As an example, we show that the spectrum is sensitive to even small perturbations in the timing and amplitudes of glottal pulses. In addition, we illustrate important characteristics that can arise in the spectrum, including apparent shifting of the harmonics and the appearance of multiple pitches. For stochastic perturbations, we arrive at a formulation of the power-spectral density as the sum of a low-pass line spectrum and a high-pass noise floor. Our findings are relevant to a number of speech-processing areas including linear-prediction analysis, sinusoidal analysis-synthesis, spectrally derived features, and the analysis of disordered voices.
READ LESS

Summary

Regions of nonmodal phonation, which exhibit deviations from uniform glottal-pulse periods and amplitudes, occur often in speech and convey information about linguistic content, speaker identity, and vocal health. Some aspects of these deviations are random, including small perturbations, known as jitter and shimmer, as well as more significant aperiodicities. Other...

READ MORE

Topic identification from audio recordings using word and phone recognition lattices

Published in:
2000 IEEE Workshop on Automatic Speech Recognition and Understanding, 9-13 December 2007, pp. 659-664.

Summary

In this paper, we investigate the problem of topic identification from audio documents using features extracted from speech recognition lattices. We are particularly interested in the difficult case where the training material is minimally annotated with only topic labels. Under this scenario, the lexical knowledge that is useful for topic identification may not be available, and automatic methods for extracting linguistic knowledge useful for distinguishing between topics must be relied upon. Towards this goal we investigate the problem of topic identification on conversational telephone speech from the Fisher corpus under a variety of increasingly difficult constraints. We contrast the performance of systems that have knowledge of the lexical units present in the audio data, against systems that rely entirely on phonetic processing.
READ LESS

Summary

In this paper, we investigate the problem of topic identification from audio documents using features extracted from speech recognition lattices. We are particularly interested in the difficult case where the training material is minimally annotated with only topic labels. Under this scenario, the lexical knowledge that is useful for topic...

READ MORE

Sinewave analysis/synthesis based on the fan-chirp transform

Published in:
Proc. IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, WASPA, 21-24 October 2007, pp. 247-250.

Summary

There have been numerous recent strides at making sinewave analysis consistent with time-varying sinewave models. This is particularly important in high-frequency speech regions where harmonic frequency modulation (FM) can be significant. One notable approach is through the Fan Chirp transform that provides a set of FM-sinewave basis functions consistent with harmonic FM. In this paper, we develop a complete sinewave analysis/synthesis system using the Fan Chirp transform. With this system we are able to obtain more accurate sinewave frequencies and phases, thus creating more accurate frequency tracks, in contrast to a system derived from the short-time Fourier transform, particularly for high-frequency regions of large-bandwidth analysis. With synthesis, we show an improvement in segmental signal-to-noise ratio with respect to waveform matching with the largest gains during rapid pitch dynamics.
READ LESS

Summary

There have been numerous recent strides at making sinewave analysis consistent with time-varying sinewave models. This is particularly important in high-frequency speech regions where harmonic frequency modulation (FM) can be significant. One notable approach is through the Fan Chirp transform that provides a set of FM-sinewave basis functions consistent with...

READ MORE