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Speech and Audio Processing, IEEE Transactions on

Issue 6 • Date Nov 1995

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Displaying Results 1 - 10 of 10
  • A subband approach to time-scale expansion of complex acoustic signals

    Publication Year: 1995 , Page(s): 515 - 519
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (464 KB)  

    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 consecu... View full abstract»

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  • A fast determination of stochastic excitation without codebook search in CELP coder

    Publication Year: 1995 , Page(s): 473 - 480
    Cited by:  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (792 KB)  

    The major drawback of the code excitation linear prediction (CELP) coder is computational complexity that finds the best excitation vector from a stochastic codebook. To provide a synthesized speech signal with reasonable quality, the size of the stochastic codebook should be large. For this reason, the search becomes highly complex. To overcome this difficulty, several methods have been proposed.... View full abstract»

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  • Neural network filters for speech enhancement

    Publication Year: 1995 , Page(s): 433 - 438
    Cited by:  Papers (7)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (660 KB)  

    In adaptive noise cancelling, linear digital filters have been used to minimize the mean squared difference between filter outputs and the desired signal. However, for non-Gaussian probability density functions of the involved signals, nonlinear filters can further reduce the mean squared difference, thereby improving the signal-to-noise ratio at the system output. This is illustrated with a two-m... View full abstract»

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  • Adaptive cepstral analysis of speech

    Publication Year: 1995 , Page(s): 481 - 489
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (704 KB)  

    This paper proposes an algorithm for adaptive cepstral analysis based on the UELS (unbiased estimation of log spectrum). In the UELS, the model spectrum is represented by cepstral coefficients and the mean square of the inverse filter output is minimized with respect to the cepstral coefficients. By introducing an instantaneous gradient estimate of the criterion in a similar manner of the LMS algo... View full abstract»

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  • Analysis of the filtered-X LMS algorithm

    Publication Year: 1995 , Page(s): 504 - 514
    Cited by:  Papers (85)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (836 KB)  

    The presence of a transfer function in the auxiliary-path following the adaptive filter and/or in the error-path, as in the case of active noise control, has been shown to generally degrade the performance of the LMS algorithm. Thus, the convergence rate is lowered, the residual power is increased, and the algorithm can even become unstable. To ensure convergence of the algorithm, the input to the... View full abstract»

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  • CELP coding using trellis-coded vector quantization of the excitation

    Publication Year: 1995 , Page(s): 464 - 472
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (948 KB)  

    We analyze the performance of a CELP coder where the vector quantization (VQ) of the excitation is replaced with trellis-coded vector quantization (TCVQ). Our results show that TCVQ performs significantly better than VQ, with reasonable complexity. This makes TCVQ a fair choice for trading quality against complexity and/or delay. We describe a systematic procedure to replace VQ with TCVQ for exist... View full abstract»

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  • A fast algorithm for computing the vocal-tract impulse response from the transfer function

    Publication Year: 1995 , Page(s): 449 - 457
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (908 KB)  

    This paper describes a fast algorithm that computes the impulse response of the vocal tract from its transfer function. First, numerical methods for computing the transfer function of a given vocal-tract configuration are briefly outlined. These methods include techniques (1) to decompose the numerator and denominator of the transfer function and (2) to efficiently determine the resonance modes of... View full abstract»

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  • A differential perceptual audio coding method with reduced bitrate requirements

    Publication Year: 1995 , Page(s): 490 - 503
    Cited by:  Papers (5)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1356 KB)  

    A new audio transform coding technique is proposed that reduces the bitrate requirements of the perceptual transform audio coders by utilizing the stationarity characteristics of the audio signals. The method detects the frames that have significant audible content and codes them in a way similar to conventional perceptual transform coders. However, when successive data frames are found to be simi... View full abstract»

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  • Isolated word recognition using Markov chain models

    Publication Year: 1995 , Page(s): 458 - 463
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (660 KB)  

    The paper describes how Markov chains may be applied to speech recognition. In this application, a spectral vector is modeled by a state of the Markov chain, and an utterance is represented by a sequence of states. The Markov chain model (MCM) offers a substantial reduction in computation, but at the expense of a significant increase in memory requirement when compared to the hidden Markov model (... View full abstract»

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  • Reduction of broad-band noise in speech by truncated QSVD

    Publication Year: 1995 , Page(s): 439 - 448
    Cited by:  Papers (66)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (948 KB)  

    We consider an algorithm for reduction of broadband noise in speech based on signal subspaces. The algorithm is formulated by means of the quotient singular value decomposition (QSVD). With this formulation, a prewhitening operation becomes an integral part of the algorithm. We demonstrate that this is essential in connection with updating issues in real-time recursive applications. We also illust... View full abstract»

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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.

Full Aims & Scope