Proceedings of the 2017 4th International Conference on Advances in Electrical Engineering (ICAEE), 28-30 September, Dhaka,

Bangladesh

Modified Mel-frequency Cepstral Coefficients

(MMFCC) in Robust Text-dependent Speaker
Identification.

Md. Atiqul Islam

Department of Electrical and Electronic Engineering
International Islamic University Chittagong
Chittagong, Bangladesh
atiq.atrai@gmail.com

Abstract: In this paper a new approach in close-set speaker
identification has been presented. The traditional mel-
frequency cepstral coefficient (MFCC) feature has been
modified and it’s name was given modified MFCC
(MMFCC). Text-dependent dataset has been used to test the
presented method’s speaker identification rate. The speaker
identification rate of the presented study was estimated in
both clean and contaminated conditions. Four types of noises
were added to clean signal to get noisy signals for a limit of
signal to noise ratios (SNRs) from -5dB to 10 dB. Moreover,
the obtained performance was compared with the
performance of the traditional features like MFCC- and
Gammatone frequency cepstral coefficient (GFCC)-based
methods. The evaluated results showed the proposed method
achieves significant improved performance over conventional
MFCC- and GFCC-based methods performance under noisy
conditions.
Index Terms: Text-dependent, robust speaker identification,
modified MFCC (MMFCC), envelope.
I. INTRODUCTION
Automatic speaker identification is belongs to the biometric
process of identifying a target speaker from a number of known
or unknown speakers by matching voice pattern against a
number of speaker models. Now a day, Speaker identification is
becoming popular in trading, banking, shopping, crime
investigation, information reservation and as a forensic tool
because of unique and distinguishing characteristics of individual
voice production mechanism. Generally, speaker identification is
done using both text-dependent and independent speech signal.
In automatic text-dependent speaker identification system,
testing speech is confined to specific utterances with several
sessions. In this paper, a new text-dependent speaker
identification approach has been studied. This process is
implemented in three steps: extraction of feature from audio
signal, speaker prototype behavioral modeling and testing the
system performance based on pattern classification process.
The feature extraction from voice signal is the most
challenging issue in automatic speaker identification. A
successful feature extraction provides speaker distinguishing
characteristics which fit properly in the speaker modeling. This
model gives robust speaker identification performance
irrespective of nature of environment. That’s why, a great
emphasized is given on extraction of feature from audio speech
signal. To obtain successful feature from voice signals, a number
of feature extraction models have been developed. In broad sense
all feature extraction models are categorized into two groups:
auditory periphery-based feature extraction model such as
(MFCC) [1] and (GFCC) [2] and voice production system based
model like perceptual linear predictive (PLP) coefficients [3] and
linear prediction cepstral coefficients (LPCCs) [4].
The MFCC is mostly used as front-end method in speaker,
speech, and phoneme classification. Now-a-days this feature is
used as a standard feature to compare the performances from
newly proposed method. MFCC provides better speaker
identification accuracy in clean condition [5, 6] but in noisy
conditions (at 0 dB SNR) it’s performance falls to a remarkable
level [7]. The application of cubic root operation in lieu of log
operation in MFCC feature can slightly improve speaker
identification accuracy under noisy conditions [8].
The study [5] shows, the inclusion of phase information
enhance MFCC performance. According to the [9] investigation,

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505
the phase spectrum and the magnitude spectrum both contribute
to speech intelligibility depending on the proper selection of
window shape. However, the voice of a sound is identified
mainly by magnitude spectra as observed by [10].
A new feature named GFCC [3] unlike MFCC in filter
(Gammatone filter rather using of triangular filter) and non-
linear operation (cubic root operation alternative of log
operation) has been proposed to obtain improved performance
over MFCC. This feature-based method provides almost similar
results of MFCC-based method for text-dependent speaker
identification system [11].
Voice is the multi-dimensional presentation and it’s not
possible to present it with a single variable. A possible approach
could be to look at sounds overall spectral energy distribution.
The importance of energy estimation from basilar membrane
has been discussed in [12]. According to this study, auditory
system averages basilar membrane energy (is called envelope)
and transmit the sound information through auditory nerve. So,
in this study the energy of filters bank responses has been
accumulated and returned to time domain applying discrete
cosine transform (DCT). This new feature is presented in this
study is called modified mel-frequency cepstral coefficient
(MMFCC).This feature has been applied through this study to
obtain robust speaker identification score using UM dataset [11].
The achieved performance was significantly improved in
contrast to the performances of the conventional MFCC- and
GFCC-based methods under noisy conditions. In summary; in
this paper a new feature modifying conventional MFCC front-
end speech feature extraction in speaker identification has been
presented. The obtained performance outperformed similar type
feature- (MFCC- and GFCC-) based performance under noisy
conditions. The proposed feature extraction algorithm also
supports physiology of human peripheral auditory mechanism.
Fig. 1: The methodological block diagram of MMFCC feature
extraction from an audio speech signal.
II. METHODOLOGY
In this section, the proposed feature and the baseline feature
extraction process from audio speech signal, experimental setup
and speaker modeling has been described.
506
A. Modified MFCC (MMFCC) Extraction
The feature extraction from audio speech signal is almost
similar to conventional MFCC. Fig. 1 represents the block
diagram for the extraction process of the newly proposed feature.
Initially, input signal was pre-emphasized to make equal-
loudness to audio waveform. Fig. 2 represents a typical speaker
identification methodological block diagram.
Most of all auditory periphery-based models have a common
process to make similarities with human auditory system. The
audio signal is transgressed through a number of band pass filters
considering basilar membrane as band pass filters bank with
overlapping pass-bands. In this study 25 channel linear spaced
band pass (same as MFCC) filters have been used. A hamming
window (25 ms) with 10 ms overlap has been addressed to
simulate basilar membrane response.
The pre-emphasized audio speech signal was passed through
hamming window to obtain a matric feature like u × v. Where u
is equal to the number of window points and v is the number of
column that fits the total signal length. It was implemented using
following formula,
݈െ‫݋‬
‫ݒ‬ൌ ǥ ǥ ǥ ǥ ǥ ǥ ǥ Ǥ ሺͳሻ
‫ݑ‬െ‫݋‬
Here, l is the length of audio speech signal and o is the number
of overlap between adjacent windows. Fast Fourier transform
was applied on the obtained matric feature. In this study, 512
points of input signal was used as number of fast Fourier
transform (NFFT). The obtained feature is called specgram is
similar to spectrogram. Then, power spectrum was taken of the
specgram. In conventional MFCC feature, log operation is used
to reflect cochlea non-linearity. Here a cubic root operation was
applied on the power spectrum to scale the loudness according to
Stevens [13-14] as
ଵ
‫ܧ‬ሺ݅ǡ ݆ሻ ൌ ܾܽ‫ ݏ‬൬‫ݕ‬ሺ݅ǡ ݆ሻଷ ൰ ǥ ǥ ǥ ǥ ǥ ǥ ǥ ሺʹሻ
The obtained power spectrum is then forwarded through
uniformly spaced filters bank. In this study, 25 filters were used
with frequency range from 0 Hz to half of sampling frequency
(as used in original MFCC feature extraction process). The
frequency components of power spectrum were converted into
Mel scale following MFCC extraction procedure.
R. S. Holambe and M. S. Deshpande investigated in [12], the
human auditory system averages the energies of each critical
band frequencies of basilar membrane and finally presents a
compressed form of the input audio signal. So, the average
energy of the filters bank responses was taken.

Fig. 2: The newly implemented method’s functional block
diagram.
In this study, 6 points of each band response was considered
as a frame with 50% overlap and energy of each band was
computed following
௧ identification process. A text-dependent dataset University
ܵሺ݅ǡ ݉ሻ ൌ ‫ݔ‬ሺ݅ǡ ͳǣ ݉ ‫ݎ כ‬ሻ ‫ݔ כ‬ሺ݅ǡ ͳǣ ݉ ‫ݎ כ‬ሻ ǥ ǥ ǥ Ǥ Ǥ ሺ͵ሻ
Here, i indicates the number of filter band. m is the number of
frames with overlap of r. t is stands for transpose of matric. This
process reduces the feature size about one-third of the
conventional feature. Finally, a discrete cosine transform was
applied to convert the obtained energy spectrum into cepstral
coefficient. The obtained feature name was given modified mel-
frequency cepstrum coefficient (MMFCC). Here is to be
mentioned that, the dynamic and acceleration coefficients were
not included in MMFCC that could be further studied topic.
B. Baseline Feature Extraction
The feature extraction processes of MFCC and GFCC have
been described below chronologically.
i. Mel-frequency cepstral coefficient (MFCC)
The feature extraction process of conventional MFCC is
almost similar to the proposed feature obtaining process. The
difference between them only in cochlea non-linearity operation:
MFCC uses only operation on FFT-based power spectrum and
does not average filter responses energies. The number of filter
bands and frequency range all parameters were kept same as the
proposed feature extraction procedure.
The MFCC feature extraction from audio speech signal was
done using rastamat toolbox [13]. To make fare comparison with
the proposed method, only static coefficients of MFCC have
been taken into consideration.
ii. Gammatone filter cepstral coefficient (GFCC)
GFCC feature is almost similar to MFCC feature. GFCC uses
Gammatone filter rather using of triangular filter to reflect
507
cochlea basilar membrane responses and applies cubic root
operation on filter responses to add cochlea non-linearity. In this
study, 64 bands of Gammatone filter have been used to simulate
frequency responses from 50 Hz to 3 kHz following [14] and
frequency range has been chosen to keep pace with the proposed
feature information. The filter responses were decimated to 100
Hz which is equivalent to framing and a cubic root operation was
implemented. Finally, DCT was applied to convert the spectral
information into time domain. It was observed in [14], the most
speech information retains in 1st to 23rd bands after applying
DCT operation due to its compaction properties. So, only first
23rd coefficients were used this study as GFCC feature. To be
mentioned, no FFT application is needed in GFCC feature
extraction.
C. Experimental Setup
This paper presents a close-set text-dependent speaker
Malaya (UM) [11] has been used in this paper to record the
speaker identification rate of the newly proposed method in clean
and noisy conditions. The noisy speech was obtained by adding
four various noises for an extent of SNRs from -5 dB to 10 dB at
step of 5dB. White, pink, street, and babble noises were used as
background noise in noisy signals.
UM dataset contains 39 speakers. Each speaker has ten speech
samples. Each sample’s utterance was ‘University Malaya’.
Seven samples from each speaker were used to create speaker
prototype behavioral model. Only clean signals were used to
train the speaker model. Once speaker model was ready, it was
saved for testing the proposed method validation. Rest three
samples were taken to test the presented method performance in
clean and distorted conditions.
D. Speaker Modeling
The most crucial task in speaker identification is modeling
speaker behavioral model. A successful classifier can extract
latent parameters from extracted feature from audio speech
signal that characterizes each individual speaker identity. The
adequate information availability ensures accurate speaker
modeling this study, Gaussian mixture model-universal
background model (GMM- UBM) [15] has been. has been used
in this study for speaker modeling to achieve robust SID
performance.
The GMM speaker modeling is adapted with the UBM-based
trained speaker data to make the system faster, stable,
and to have better performances. The application of expectation
maximization (EM) [16] in GMM-based speaker modeling
makes it successful classifier. The positive sides of application of
EM is that it can capture required latent parameters for GMM
from a small quantity of training data, and the obtained White Noise
parameters can be used to the new data by maximum a-posteriori 100
(MAP) adaptation [17].
A GMM-UBM-based classifier with 128 mixture components 80

SID Accuracy (%)

has been used here to train the newly developed features to obtain
speaker prototype behavioral model for each speaker. The same 60
algorithm has been also run for the MFCC- and GFCC-based
methods to make fair comparison. 40

III. RESULT AND ANALYTICAL STUDY 20 MMFCC

MFCC
In this section, the proposed method-based text-dependent GFCC
0
result has been presented. The proposed method-based Pink noise
performance was evaluated using both clean and noisy signal. 100
The obtained performance has been compared with the
conventional MFCC- and GFCC-based methods performance to 80

SID Accuracy (%)

show the novelty of the proposed feature.
60
Fig. 3 represents the speaker identification rate of MMFCC,
MFCC-, and GFCC-based method. The common thing in speech
40
processing is the automatic identification rate reduces with the
increment of noise level. It is seen from Fig. 3, the MFCC- and MMFCC
20
GFCC-based performances were dropped continuously with the MFCC
GFCC
positive change of noise level. The performance decrement rate 0
for baseline methods was almost linear with the increment of Street noise
SNR level. However, the proposed MMFCC-based performance 100
was comparatively robust to noises than the performance of other
baseline methods. SID Accuracy (%) 80
The obtained result also implies the simulation-based method
provides better speaker identification accuracy for slow varying 60
noise like pink noise and mostly suffered for stationary noises.
The MMFCC-based method’s performance was more than 40% 40
irrespective of noise except street noise at 0 dB SNR. On the MMFCC
other hand, the baseline methods performances were less than 20 MFCC
GFCC
20% at 0 dB SNR. The GFCC-based method provides
0
comparatively better result than MFCC for all noises except
Babble noise
babble noise irrespective of SNRs which is consisted with the 100
result of [11]. On the other side, MFCC generates better
identification score over GFCC for babble noise under noisy 80
SID Accuracy (%)

condition. In summary, it can be said based on the Fig. 3 result;

the newly proposed method provides significantly improved 60
performance over other existing methods irrespective of noises
and SNRs. 40
To investigate the robustness issue, the proposed method was
run without application of DCT. It was observed, the 20 MMFCC
MFCC
performance was 100% in clean but it reduced to a significant GFCC

level under noisy conditions compared to presented results. So, it 0

can be apparently said based on the obtained result; the human -5 0 5 10 Clean
auditory system finally converts spectral information of audio SNR (dB)
speech into time domain to identify a target speaker in noisy
environment. Fig. 3: Comparison of text-dependent speaker identification
There have a number of nonlinearities in human auditory results for four various noises for a range of SNRs using the
proposed (MMFCC-), MFCC- and GFCC-based methods.

508
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