Moroccan Dialect Speech Recognition System Based

on CMU SphinxTools
Abderrahim Ezzine
LISAC Laboratory
Faculty of Sciences Dhar
Mahraz, Sidi Mohammed Ben
Abbdallah University
Fés, Morocco
ezzine2abderrahim@gmail.com
Hassan Satori
LISAC Laboratory
Faculty of Sciences Dhar
Mahraz, Sidi Mohammed Ben
Abbdallah University
Fés, Morocco
hassan.satori@usmba.ac.ma
Mohamed Hamidi
LISAC Laboratory
Faculty of Sciences Dhar
Mahraz, Sidi Mohammed Ben
Abbdallah University
Fés, Morocco
mohamed.hamidi.5@gmail.com
Khalid Satori
LISAC Laboratory
Faculty of Sciences Dhar
Mahraz, Sidi Mohammed Ben
Abbdallah University
Fés, Morocco
khalidsatori@gmail.com

Abstract— The main aim of an Automatic Speech Recognition
system (ASR) is to produce a system that is able to simulate the
human listener based on the learning approach and speech data of
a studied language. In this paper, we describe the Darija
Moroccan Dialect speech recognition system that is implemented
to recognize the ten first Arabic digits spoken in Moroccan dialect
(Darija) collected from 20 speakers including both males and
females. This system is designed based on the CMU Sphinx tools
through the ASR Hidden Markov Model method with small data
and the Mel frequency spectral coefficients (MFCCs) that are used
in the feature extraction phase. Our best-obtained accuracy is
96.27 % found with 8 GMMs.
Keywords—Speech recognition; Moroccan dialect; HMMs;
MFCC; CMU Sphinx; Acoustic model; Artificial intelligence.
I. INTRODUCTION
Automatic Speech Recognition (ASR) defined as a
technology that allows a computing device to converts the words
into a readable text by way of a microphone or telephone. The
ASR has a large field of implementations such as command
recognition, interactive voice response, dictation, it can be used
to help handicapped people to interact with the community. It is
a technology that makes life more facile [1]. The principal aim
of ASR research is to permit a computer to identifies all words
that were spoken by anybody, independent of vocabulary size,
noise and speaker characteristics in real-time with 100%
precision [2].
To build an ASR system, we need to create the Language
Model (LM), Acoustic Model (AM), and dictionary for the
target language. Unfortunately, designing an acoustic model for
a specific language is expensive, unlike the AM and the
dictionary. Due to the recording of speech data from speakers to
ensure the ASR speaker-independent [3]. Given the importance
of ASR technology, several systems are implemented for
different languages based on the HTK [4] and CMU Sphinx [5],
Dragon [6], KALDI [7] toolkits.
H. Satori and F. ElHaoussi [8] have implemented an
Amazigh speech system by using CMU Sphinx tools based on
Hidden Markov Model. The exploited corpus includes 60
Amazigh Moroccan speakers Tarifit native, equally divided
between male and female. Their designed system permits to
recognize digits and alphabets of Amazigh language and the
best-achieved performance is 92.89% found with 16 Gaussian
Mixture models.
Ouissam et al. [9] have presented an automatic speech
recognition system based on Sphinx4 that permit to detect the
people who have disorders voices. Their project is carried out
using Amazigh language in order to differentiate the normal and
pathological voices. Their findings were measured using
combinations of HMMs 5-states with 8 Gaussian mixture
distributions.
Hamidi et al. [10] have presented an interactive security
system-based speech recognition technique. In their work, the
ASR-HMM and IVR technologies were combined to allow the
distance tasks administration managing by utilizing speech
commands and the security identification by using biological
voiceprint. The findings present that the access rate is more than
80 % whereas the non-admin recognition rate is less than 6%.
The aim of this paper is to create the Darija dialect ASR and
explore the changes that must be realized in the model to adapt
Moroccan dialect speech recognition. Our work will be based on
the hidden Markov model - Gaussian mixture model
combination. The proposed system will be designed by using
Carnegie Melon University (CMU) Sphinx which is a statistical
speaker-independent set of tools using the Hidden Markov
Models (HMM).
The paper is organized as follows: Section 2 presents the
Moroccan dialect. Section 3 presents the Moroccan dialect
speech recognition system. Section 4 shows the experimental
results and Section 5 concludes the paper.

978-1-7281-8041-0/20/$31.00 ©2020 IEEE

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 II. MOROCCAN DIALECT TABLE I. TEN FIRST DIGITS WITH THEIR SYLLABLES AND THEIR
TRANSCRIPTION IN ENGLISH, ARABIC

In this paper, we studied a Moroccan dialect called Darija is a

spoken variety of Arabic. It is considered the informal language
Moroccan
of Morocco and it utilized in daily communication, media like English
Digits Dialect
TV and Radio. Given its direct contact with the Amazigh transcription
transcription
language, the Moroccan dialect is much influenced by the 0 SIFR 
Amazigh sound pattern, morphology and lexicon. Moroccan
1 WAHD 
dialect has 32 basic phonemes, of which four are vowels, and 28
are consonants. Moroccan dialect has fewer vowels than 2 JOJ

classical Arabic [11]. 3 THLATA  

III. MOROCCAN DIALECT SPEECH RECOGNITION SYSTEM 4 RABAA 

5 KHAMSA 
This part represents our experience to build and develop a 6 STTA 
Moroccan dialect voice recognition system using CMU Tools.
7 SBAA 
Fig. 1 represents the main elements that are usually found in a
typical ASR system. 8 THMANYA 
Recently, the applications of automatic speech recognition for 9 TSAAOD 

Moroccan Amazigh language based on CMU Sphinx tools was
targeted by our lab researchers [12-18].
B. Corpus preparation

The first step of ASR system construction is preparing the

corpus which is a collection of unit sounds defined by certain
words in the vocabulary. The corpus was created by using the
Text ten 10 Darija Moroccan dialect digits. A number of 20 Moroccan
Corpus speakers (10 males and 10 females) aged between 14 and 50
Feature years old were asked to utter all digits 10 times. Hence, the
Extraction database Moroccan Digit consists of 10 repetitions of every digit
produced by each speaker. Depending on this, the database for
Moroccan speakers consists of 2000 tokens. During the
Language recording session, each utterance was played back to assure that
Model the entire digit was included in the registered signal. Table II
Decoder shown the parameters used for the corpus preparation.
Speech Acoustic
Model TABLE II. CORPUS PARAMETERS
Corpus

Parameter Value
Text
Sampling rate 16 kHz
Fig. 1. Block diagram of ASR System
Number of bits 16 bits

A. System overview Wave format Mono, wav

Corpus 10 Moroccan Arabic-digits

The main aim of our study is to design a Darija automatic speech Repetitions 10 times
recognition system based on Mel frequency spectral coefficients
in the feature extraction phase and GMM-HMM system Pronunciation Darija Moroccan Dialect
combination techniques in the training phase. Condition of noise Normal life
To build our Darija acoustic model we have used SphinxTrain
tools based on the dictionary, language Model and collected Speakers 10 male 10 female
speech data while in the recognition phase we have based on
Pocketsphinx decoder. Our adopted dictionary file was created C. Training
by using the ten first Darija Moroccan dialect digits followed by
their transcriptions. Table I represents the Moroccan the first The Training of acoustic model is done using CMU Sphinx
Darija digits with their pronunciation. tools that uses embedded training method based on the Baum-

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Welch algorithm. Training is the procedure of building the
knowledge base by learning the Acoustic Model and Language Pre-
Framing
Model used by the system [8]. Fig. 2 presents the training emphasis
process.
The generation of the acoustic model is realized by grouping
a set of input data and treat them with the SphinxTrain tool (see
Fig. 8). The input data is as follow: Mel frequency
FFT Windowing
• Speech data filtering

• Transcript files that include the word pronounced set

for each wave file.
• Fileids files that include the path of each wave file.
• Dictionary and Language Model files Log 10 DCT Acoustic
Vectors
• Phone file that lists the used phonemes set.
Fig. 3. MFCC Architecture

Training The extraction of Mel-Frequency Cepstral Coefficients

transcription Testing transcription (MFCC) includes an analysis based on the frames of an input
And fileids files And fileids files speech where the speech signal is segmented into a sequence of
frames. Each frame offers a sinusoidal transformation (Fast
Fourier Transform) to generate certain parameters, which then
Dictionary undergoes a perception scale on the Mel scale and a
and lm.DMP decorrelation. The output obtained was a sequence of
files characteristic vectors describing a logarithmically useful
compressed amplitude and simplified frequency information.
Configuratio
n
Phone and file
2) Acoustic model
filler files
MDdigits

Among the most defies of automatic speech recognition is

Training files
Wave files
Testing files
Fig. 2. Proccess of training phase
1) Feature extraction
Feature extraction technique transform the speech waveform
into a sequence of feature vectors which contains only the
required information to identify a given utterance where play
important role in speech recognition system performance. The
most widely used methods of a feature vectors extraction is Mel-
Frequency Cepstral Coefficients (MFCC), which is used to
simulate the human ear. Fig. 3 gives a summary of the extraction
of Mel coefficients (MFCC) [19].
the accuracy. The acoustic model plays an important role for
ameliorate this accuracy. The main goal of the acoustic model is
to compute the likelihood of the observed feature vectors given
linguistic units (phones, words, subparts of phones) using a
statistical method known as the Hidden Markov Model (HMM)
with a mixture density Gaussian distribution. For example,
Gaussian Mixture Model (GMM) is used to calculate the
likelihood of a given feature vector P (O | Q) for each HMM
state Q, corresponding to O phone or subphone. For recognizing
small numbers of words like 10 digits, using HMM state to
represent a phone is sufficient. the popular configuration to
represent the phone is three HMM states, each phone has three
emitting HMM states instead of one plus two non-emitting states
at two ends. This 5 states phone HMM is known as a word model
or phone model as shown in fig. 4.
3) Language model
The language model (LM) is utilized by the speech
recognition system to guide the search for correct word
sequences. There are many types of models to describe any
language in the recognition and recognition phases such as
grammar, phonetics statistical language models and statistical
language models. In this work, for generate the Language model

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of our system we use the CMU-Cambridge statistical language TABLE III. RECOGNITION RATES TO THE TEN DARIJA DIGITS WITH THREE
HMM STATES AND DIFFERENT GMMS VALUES
modelling toolkit [20].

GMMs
S Moroccan Dialect
digits
4 8 16 32

1 2 3 SIFR 91.66% 95.00% 91.66% 90.00%

WAHD 96.66% 96.66% 95.00% 95.00%

(a) JOJ 96.66% 93.33% 95.00% 95.00%

THLATA 95.00% 95.00% 93.33% 91.66%

S TT A RABAA 93.33% 90.00% 88.00% 90.00%

KHAMSA 86.66% 91.66% 91.66% 86.00%

1 2 3 1 2 1
3 2 3
STTA 76.66% 81.66% 85.00% 75.00%

SBAA 95.00% 93.33% 93.33% 90.00%

(b) THMANYA 96.66% 95.00% 93.33% 93.33%

Fig. 4. (a) representation of "S" Phoneme with 5 hmm states (b) TSAAOD 96.66% 93.33% 91.66% 93.33%
representation of “STTA” digits with hmm states.
Average 95.08% 96.27% 96.10% 94.58%

4) Pronunciation dictionary
TABLE IV. RECOGNITION RATES TO THE TEN DARIJA DIGITS WITH FIVE
The dictionary file is used as an intermediary among the AM HMM STATES AND DIFFERENT GMMS VALUES
and LM. Our used pronunciation dictionary file includes the ten
first Darija dialect words followed by their pronunciation. Fig. 5 GMMs
Moroccan Dialect
represents the phonetic dictionary list used in the training [21]. digits
4 8 16 32

SIFR SS I F R SIFR 98.33% 95.00% 93.33% 81.66%

WAHD W A HH D WAHD 98.33% 96.66% 93.33% 95.00%
JOJ J OU J
THLATA TH L A TH A JOJ 98.33% 98.33% 96.66% 96.66%
RABAA R A B AA A THLATA 98.33% 96.66% 96.66% 95.00%
KHAMSA KH A M S A
STTA S TT A RABAA 91.66% 96.66% 80.00% 80.00%
STTA(2) S E TT A KHAMSA 88.33% 93.33% 90.00% 91.66%
SBAA S A B AA A
THMANYA TH M A N Y A STTA 75.00% 81.66% 83.33% 85.00%
TSAAOD T S AA OU D SBAA 95.00% 93.33% 96.66% 98.33%
TSAAOD(2) T E S AA D
THMANYA 98.33% 96.66% 95.00% 96.66%
Fig. 5. The used phonetic dictionary list. TSAAOD 96.66% 96.66% 93.33% 95.00%

IV. EXPERIMENTAL RESULTS Average 95.25% 95.25% 94.92% 92.71%

In all proposed schemes, the used data were partitioned to 70

% for training and 30 % for testing to assure the speaker-
independent aspect. Also, our system was trained by exploiting
various GMMs values that ranged from 4 to 32 combined with
3 and 5 states per HMMs.
In the case of the three HMM states, Table IV presents the
systems recognition rates as well as the average recognition rates
of all digits. The system recognizes 2000 token for all ten digits.
The obtained system performances are 95.08 % 96.27 % 96.10
% and 94.58 % were found by utilizing 4, 8, 16 and 32 GMMs,

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respectively. We have observed that 8 GMMs gives the best rate [11] M. Ennaji, A. Makhoukh, H. Es-saidy, M. Moubtassime, S. Slaoui “ A
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When we used the five HMM state, the systems try to recognize Vocal parameters analysis of smoker using Amazigh language.
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