Issue 2 • Mar 1995
Cited by: Papers (28) | Patents (2)
Several neural network-based tone recognition schemes for continuous Mandarin speech are discussed. A basic MLP tone recognizer using recognition features extracted from the processing syllable is first introduced. Then, some additional features extracted from neighboring syllables are added to compensate for the coarticulation effect. It is then further improved to compensate For the effect of sa... View full abstract»
A comparative study of robust linear predictive analysis methods with applications to speaker identificationPublication Year: 1995, Page(s):117 - 125
Cited by: Papers (23)
Various linear predictive (LP) analysis methods are studied and compared from the points of view of robustness to noise and of application to speaker identification. The key to the success of the LP techniques is in separating the vocal tract information from the pitch information present in a speech signal even under noisy conditions. In addition to considering the conventional, one-shot weighted... View full abstract»
Cited by: Papers (5)
A generalized minimal distortion segmentation algorithm is proposed to solve the time alignment problem for ANN-based speech recognition. By modeling dynamics of spectral information of an acoustic segment with smooth curves obtained by orthonormal polynomial expansion, a speech signal is optimally divided into segments and then recognized by an MLP recognizer. Experimental results showed that the... View full abstract»
Cited by: Papers (5)
Adaptive quantization is used in many ADPCM systems to improve performance. The authors present an analysis dealing with the effect of adaptive quantizers on the stability of ADPCM systems as compared with ADPCM with nonadaptive quantization. Stability of the system is noted to equate with recovery from encoder internal state discrepancies. A sufficient condition for stability that leads to a rela... View full abstract»
Cited by: Papers (55) | Patents (1)
In most acoustic echo canceler (AEC) applications, an adaptive finite impulse response (FIR) filter is employed with coefficients that are computed using the LMS algorithm. The paper establishes a theoretical basis for the slow asymptotic convergence that is often noted in practice for such applications. The analytical approach expresses the mean-square error trajectory in terms of eigenmodes and ... View full abstract»
Aims & Scope
Covers the sciences, technologies and applications relating to the analysis, coding, enhancement, recognition and synthesis of audio, music, speech and language.
This Transactions ceased publication in 2005. The current retitled publication is IEEE/ACM Transactions on Audio, Speech, and Language Processing.