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  • Conference paper
    Moore AH, Naylor PA, Skoglund J, 2014,

    An Analysis of the Effect of Larynx-Synchronous Averaging on Dereverberation of Voiced Speech

    , European Signal Processing Conference, ISSN: 2219-5491
  • Conference paper
    Stanton R, Gaubitch N, Naylor P, Brookes DMet al., 2014,

    A Differentiable Approximation to Speech Intelligibility Index with Applications to Listening Enhancement

    , AES Intl Conf on Audio Forensics

    The Speech Intelligibility Index is a standardised objective measure for estimating the intelligibility of speech in noise. It is, however difficult to use it in the iterative optimisation of speech enhancement algorithms because it is a discontinuous function of its input parameters. In this paper, we derive an approximation for the Speech Intelligibility Index that is both continuous and differentiable, which allows for more efficient optimisation procedures. The use of the approximation is demonstrated in an application to near-end speech enhancement.
  • Conference paper
    Moore AH, Brookes M, Naylor PA, 2013,

    Roomprints for forensic audio applications

    , Proc. IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), Publisher: IEEE

    A roomprint is a quantifiable description of an acoustic environment which can be measured under controlled conditions and estimated from a monophonic recording made in that space. We here identify the properties required of a roomprint in forensic audio applications and review the observable characteristics of a room that, when extracted from recordings, could form the basis of a roomprint. Frequency-dependent reverberation time is investigated as a promising characteristic and used in a room identification experiment giving correct identification in 96% of trials.
  • Journal article
    Gaubitch N, Brookes M, Naylor P, 2013,

    Blind Channel Magnitude Response Estimation in Speech using Spectrum Classification

    , IEEE Transactions on Audio, Speech, and Language Processing, Vol: 21, Pages: 2162-2171, ISSN: 1558-7916
  • Conference paper
    Moore AH, Brookes M, Naylor PA, 2013,

    Room geometry estimation from a single channel acoustic impulse response

    , Proc. European Signal Processing Conference (EUSIPCO)
  • Conference paper
    Eaton D, Brookes DM, Naylor PA, 2013,

    A Comparison of Non-Intrusive SNR Estimation Algorithms and the Use of Mapping Functions

    , EUSIPCO, Publisher: EURASIP, Pages: 1-5

    We present a comparative evaluation of six methods for non-intrusive Signal-to-Noise Ratio (SNR) estimation for narrowband speech in noise. We demonstrate that the performance of all methods can be improved by applying a non-linear mapping function to their estimates of SNR. We have employed phrases built from the TIMIT speech corpus and noises from a broad range of sources including ITU-T P.501, NOISEX-92, and Soundjay. We compare the accuracy of the methods in estimating the SNR of both stationary and non-stationary noise and we conclude that with the mapping function, the best current methods can estimate the SNR to within approximately 3.5 dB for SNRs from -5 dB to 35 dB.
  • Conference paper
    Sharma D, Naylor PA, Brookes M, 2013,

    NON-INTRUSIVE SPEECH INTELLIGIBILITY ASSESSMENT

    , 21st European Signal Processing Conference (EUSIPCO), Publisher: IEEE
  • Conference paper
    Gaubitch ND, Löllmann HW, Jeub M, Falk TH, Naylor PA, Vary P, Brookes Met al., 2012,

    Performance comparison of algorithms for blind reverberation time estimation from speech

    The reverberation time, T<inf>60</inf>, is one of the key parameters used to quantify room acoustics. It can provide information about the quality and intelligibility of speech recorded in a reverberant environment, and it can be used to increase robustness to reverberation of speech processing algorithms. T<inf>60</inf> can be determined directly from a measurement of the acoustic impulse response, but in situations where this is unavailable it must be estimated blindly from reverberant speech. In this contribution, we provide a study of three state-of-the-art methods for blind T<inf>60</inf> estimation. Experimental results with a large number of talkers, simulated and measured acoustic impulse responses, and various levels of additive white Gaussian noise are presented. The relative merits of the three methods in terms of computational time, estimation accuracy, noise sensitivity and inter-talker variance are discussed. In general, all three methods are able to estimate the reverberation time to within 0.2 s for T<inf>60</inf> ≤ 0.8 s and SNR ≥ 30 dB, while increasing the noise level causes overestimation. The relative computational speed of the three methods is also assessed.
  • Conference paper
    Sharma D, Hilkhuysen G, Naylor PA, Gaubitch ND, Huckvale M, Brookes Met al., 2012,

    Descriptive Vocabulary Development for Degraded Speech

    , 13th Annual Conference of the International-Speech-Communication-Association, Publisher: ISCA-INT SPEECH COMMUNICATION ASSOC, Pages: 1494-1497
  • Conference paper
    Sharma D, Naylor PA, Gaubitch ND, Brookes Met al., 2012,

    NON INTRUSIVE CODEC IDENTIFICATION ALGORITHM

    , IEEE International Conference on Acoustics, Speech and Signal Processing, Publisher: IEEE, Pages: 4477-4480, ISSN: 1520-6149
  • Conference paper
    Sharma D, Hilkhuysen G, Gaubitch ND, Brookes M, Naylor PAet al., 2011,

    C-Qual - A validation of PESQ using degradations encountered in forensic and law enforcement audio

    , Pages: 177-181

    Assessment of speech quality of law-enforcement audio recordings is important as degradations introduced by non-ideal recording conditions can reduce the intelligence value of such recordings. Furthermore a model that predicts speech quality could be beneficial for assessing the performance of audio collection and enhancement systems. The Perceptual Evaluation of Speech Quality (PESQ) algorithm (ITU-T P.862) has been validated for degradations common in telecommunications. In this paper we apply PESQ to degradations typically encountered in law-enforcement. Also we present a subjectively labeled database (C-Qual) containing distortions encountered in law enforcement scenarios. Comparing the prediction by PESQ and the observed opinions provided by the listeners shows that PESQ is less suitable for estimating the speech quality in this context.
  • Conference paper
    Gaubitch ND, Brookes M, Naylor PA, Sharma Det al., 2011,

    Bayesian Adaptive method for estimating Speech Intelligibility in noise

    , Pages: 169-174

    We present the Bayesian Adaptive Speech Intelligibility Estimation (BASIE) method - a tool for rapid estimation of a given speech reception threshold (SRT) and the slope at that threshold of multiple psychometric functions for speech intelligibility in noise. The core of this tool is an adaptive Bayesian procedure, which adjusts the signal-to-noise ratio at each subsequent stimulus such that the expected variance of the threshold and slope estimates are minimised. Simulation results show that the algorithm is able to achieve SRT estimates accurate to within ±1 dB in under 30 iterations. Furthermore, we discuss strategies for using BASIE to evaluate the effects of speech processing algorithms on intelligibility and we give two illustrative examples for different noise reduction methods with supporting listening experiments.
  • Conference paper
    Sharma D, Naylor PA, Gaubitch ND, Miet al., 2011,

    Short-Time Objective Assessment of Speech Quality
  • Conference paper
    Gaubitch ND, Brookes M, Naylor PA, Sharma Det al., 2011,

    Single-Microphone Blind Channel Identification in Speech Using Spectrum Classification
  • Conference paper
    Sharma D, Hilkhuysen G, Gaubitch ND, Naylor PAet al., 2010,

    Data Driven Method for Non-Intrusive Speech Intelligibility Estimation
  • Conference paper
    Gaubitch ND, Brookes M, Naylor PA, 2009,

    Blind Channel Identification in Speech using the Long-term Average Speech Spectrum
  • Conference paper
    Khong AWH, Gan W-S, Naylor PA, Brookes DMet al., 2008,

    A low complexity fast converging partial update adaptive algorithm employing variable step-size for acoustic echo cancellation

    , Pages: 237-240-237-240

    Partial update adaptive algorithms have been proposed as a means of reducing complexity for adaptive filtering. The MMax tap-selection is one of the most popular tap-selection algorithms. It is well known that the performance of such partial update a.....
  • Conference paper
    Naylor P A, Khong, W H A, Brookes D Met al., 2007,

    Misalignment Performance Of Selective Tap Adaptive Algorithms For System Identification Of Time-Varying Unknown Systems
  • Journal article
    Naylor P A, Kounoudes A, Gudnason J, Brookes Met al., 2007,

    Estimation of glottal closure instants in voiced speech using the DYPSA algorithm

    , IEEE Transactions on Audio, Speech and Language Processing, Vol: 15, Pages: 34-43, ISSN: 1558-7916
  • Conference paper
    Bhunjun V, Brookes D M, Naylor P A, 2006,

    Model-based Eigen-spectrum Estimation for Speech Enhancement

    , 40th Asilomar Conference on Signals, Systems and Computers

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