- Open Access
A bio-inspired feature extraction for robust speech recognition
© Zouhir and Ouni; licensee Springer. 2014
- Received: 29 August 2014
- Accepted: 24 October 2014
- Published: 4 November 2014
In this paper, a feature extraction method for robust speech recognition in noisy environments is proposed. The proposed method is motivated by a biologically inspired auditory model which simulates the outer/middle ear filtering by a low-pass filter and the spectral behaviour of the cochlea by the Gammachirp auditory filterbank (GcFB). The speech recognition performance of our method is tested on speech signals corrupted by real-world noises. The evaluation results show that the proposed method gives better recognition rates compared to the classic techniques such as Perceptual Linear Prediction (PLP), Linear Predictive Coding (LPC), Linear Prediction Cepstral coefficients (LPCC) and Mel Frequency Cepstral Coefficients (MFCC). The used recognition system is based on the Hidden Markov Models with continuous Gaussian Mixture densities (HMM-GM).
- Auditory filter model
- Feature extraction
- Hidden Markov Models
- Noisy speech recognition
The Automatic speech recognition (ASR) system is one of the leading technologies acting on man–machine communication in real-world applications (Furui 2010). The ASR system is composed of two main modules. The first one is the acoustic Front-end (or feature extractor). This module generally uses the classical acoustic feature extraction techniques such as Perceptual Linear Prediction (PLP) (Hermansky 1990), Linear Prediction Coding (LPC) (Atal and Hanauer 1971), Linear Prediction Cepstral Coefficients (LPCC) (Atal 1974) and Mel Frequency Cepstral Coefficients (MFCC) (Davis and Mermelstein 1980). The second module is the classifier which is commonly based on the Hidden Markov Models.
The early feature based techniques involve incorporation of different psychoacoustic and neurophysical knowledge obtained from the study of the human auditory system which is capable of segmenting, localizing, and recognizing speech signal in noisy conditions without a noticeable degradation in performance of recognition (Rabiner and Juang 1993).
Generally, the feature extraction techniques are based on auditory filter modelling which uses a filterbank to simulate the cochlear filtering (Meddis et al. 2010). The efficient modelling of this auditory filterbank will improve the recognition performance and the features robustness in noisy environments.
The gammatone filterbank has been employed as the auditory filter modelling in various speech processing systems such as the Computational Auditory Scene Analysis system (Wang and Brown 2006).
Irino and Patterson have proposed an excellent candidate model for asymmetric, level-dependent cochlear filter called the Gammachirp auditory filter consistent with basic physiological data (Irino and Patterson 1997, 2006). This filter represents an extension of the gammatone filter characterized by an additional chirp parameter in order to produce an asymmetric amplitude spectrum. It provides an approximation of the auditory frequency response.
In this paper, we propose a biologically-inspired feature extraction method for robust recognition of noisy speech signals. The proposed method is based on the human auditory system characteristics, and relies on both the outer and middle ear filtering and the spectral behaviour of the cochlea. The outer and middle ear filtering is modelled by a second-order low-pass filter (Martens and Van Immerseel 1990; Van Immerseel and Martens 1992). The cochlear filter is modelled by a gammachirp auditory filterbank consisting of 34 filters, where the centre frequencies are equally spaced on the ERB-rate scale from 50 Hz to 8 kHz.
The HTK 3.4.1 toolkit is exploited in the Model training and recognition of speech signals. It is based on Gaussian Mixture density Hidden Markov models (Young et al. 2009). In our work, the HMM is trained for each word with five observation states and each state emission density consists of the four Gaussian Mixture densities.
The recognition performance of our feature extraction method was evaluated on speech signals corrupted by real-world noisy environments. The obtained results are compared to those obtained using PLP, LPC, LPCC and MFCC.
The paper is organized as follows: After introduction, section 2 presents the speech recognition system based on the hidden Markov models. It also introduces the classic feature extraction techniques of speech signals. In section 3, the proposed feature extraction method based on an auditory filter model is detailed, while introducing the auditory filter modelling. The experimental and evaluation results of our method are discussed in the section 4. Finally, conclusions are presented in the last section.
The HMM based ASR
The HMM represents a finite state machine which generates, at each state change, an acoustic vector o t observed from the probability density b j (o t ). The changes of state occur at every time unit according to the state transition probability from state i to state j is given by a ij . Figure 2 shows an example representing the observation sequence o1 to o5 for the state sequence S = 1, 2, 2, 3, 4, 4, 5, generated from a five state HMM with non-emitting entry and exit states. The HMM supports continuous Gaussian Mixture density distributions.
Where n is the dimensionality of o, ϑ is covariance matrix and μ is mean vector.
Classical feature extraction techniques
The most common techniques of feature extraction for speech recognition system employ the cepstral analysis to extract the feature coefficients from acoustic signal such as the MFCC and the LPCC. The MFCC technique consists to calculate the feature vectors from the frequency spectra at each frame of windowed speech. It is based on the human ear scale known the Mel scale.
The MFCC coefficients are calculated by applying a cosine transform to the real logarithm of short-term energy spectrum which has been expressed on a Mel-frequency scale.
Where p, G and a k are respectively the number of poles, the filter gain and the poles parameters which are called Linear Prediction Coefficients. The linear prediction coefficients are evaluated using the autocorrelation method.
The proposed extraction method of speech feature for ASR is based on an auditory filter model. This model simulates the outer/middle ear filtering and the spectral behaviour of the cochlea.
Auditory filter modelling
The auditory filter modelling represents the mathematical model which tends to simulate the basic perceptual and psychophysical aspects of the human auditory characteristics (Lyon et al. 2010). This model consists of the simulation of the outer/middle ear filtering by second-order low-pass filter and the cochlea spectral behaviour by the gammachirp auditory filterbank.
Where time t > 0, a, f0, φ and c are the amplitude, the asymptotic frequency, the initial phase and the chirp rate respectively. b and n are the two parameters which define the gamma distribution envelope. “ln” denotes the natural logarithm.
Where and Γ(n + jc) is the complex gamma distribution.
The proposed feature extraction method
This section evaluates the robustness of the proposed feature extraction method under various types of noisy environments.
Databases and experimental setup
The used speech recognition system is based on Hidden Markov Models. Our system employs the HTK 3.4.1 (Young et al. 2009) in the recognition task. The HTK 3.4.1 is a portable toolkit which allows the construction and manipulation of HMM-GM.
The HMM topology used in our experiments is a five states left-to-right model with a four Gaussian Mixture observation probability density distribution characterized by a diagonal covariance matrix.
Used Gammachirp parameters
Results and discussion
For the baseline experiments, 12 coefficients of each technique were calculated from speech signal using Hamming analysis window with length equal to 25 ms and shifted with 10 ms steps.
The recognition performance of our feature extraction method has been compared to that of the classic techniques such as PLP, LPCC, LPC, and MFCC. The feature coefficients of each technique are combined with energy (E), differential coefficients first (∆) and second order (A) (12 coefficients +E + ∆ + A).
Recognition rate (%) obtained by proposed and standard methods with suburban train noise
Recognition rate with HMM-4-GM
Suburban train noise
Recognition rate (%) obtained by proposed and standard methods with exhibition hall noise
Recognition rate with HMM-4-GM
Exhibition hall noise
Recognition rate (%) obtained by proposed and standard methods with street noise
Recognition rate with HMM-4-GM
Recognition rate (%) obtained by proposed and standard methods with car noise
Recognition rate with HMM-4-GM
As illustrated in the tables, the PLPaGc feature outperforms the four classic features in all noise conditions. For example, in the case of suburban train noise, the average of all noise levels of recognition rates achieved using PLPaGc feature is 75.47, while PLP, LPCC, LPC and MFCC feature provides respectively 65.64, 57.64, 22.97 and 65.39. It can be also observed that the recognition rates increase in all features when the noise level is decreased with respect to the signal level (i.e., SNR increases from 0 dB to 20 dB).
A new auditory filter modelling-based feature extraction method for noisy speech recognition was presented in this paper. The proposed method was motivated by the research studies of the human peripheral auditory modelling. The used auditory model consists of simulating the outer/middle ear filtering by a second order low-pass filter and the cochlea spectral behaviour by the gammachirp auditory filterbank, where the values of those centre frequencies are chosen according to the ERB rate scale. The robustness of the proposed PLPaGc feature was evaluated on speech recognition rate in real-world noisy environments. The experimental results show that the PLPaGc feature gives better recognition rates compared to four classical PLP, LPCC, LPC and MFCC feature.
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