2020 IEEE International Conference on Systems, Man, and Cybernetics (SMC)
October 11-14, 2020. Toronto, Canada
Sub-word Based End-to-End Speech Recognition
for an Under-Resourced Language: Amharic
Nirayo Hailu Gebreegziabher
Faculty of Computer Science, Data and
Knowledge Engineering Group,
Otto-von-Guericke University of
Magdeburg
Magdeburg, Germany
nirayo.hailugebreegziabher@ovgu.de
Abstract—In this work, we focused on end-to-end speech
recognition for less-resourced language, Amharic. The result can
be integrated with other tasks such as spoken content retrieval.
We explored three models, which consist of Convolutional Neu-
ral Networks, Recurrent Neural Networks, and Connectionist
Temporal Classification, towards end-to-end speech recognition
on less-resourced language. Further, we studied the possibility
of having an end-to-end system with 1-best output keeping the
network parameters and computational resource minimal. The
paper gives attention to finding a more suitable sub-lexical unit
for the Amharic end-to-end speech recognition system which
can be used as an audio indexing unit. We present the first
result comparing grapheme, phoneme, and syllable-based end-
to-end speech recognition systems for our target language. The
models are evaluated on approximately 52 hours of Amharic
speech corpus containing read-speech, audiobooks, and multi-
genre radio programs. On the test set, we report a character
error rate (CER) of 19.21% and a syllable error rate (SER) of
39.98% for a syllable-based end-to-end model without lexicons
and language model integrated.
Index Terms—Speech Recognition, Phoneme, Grapheme, End-
to-End Mode, Syllable Units.
I. Introduction
To come up with an enhanced Automatic speech recognition
(ASR) system, researchers have been using the hidden Markov
and Gaussian mixture model (HMM-GMM) paradigm for
acoustic modeling. Recently, end-to-end based architectures
covering acoustic, pronunciation, and language models in one
model have become increasingly popular and an alternative to
the hybrid architecture [1], [2].
The hybrid model, Deep Neural Network (DNN) integrated
with the HMM model, has shown significant gains over the
HMM models [3]. We have investigated the hybrid model
on the same corpus in our previous work [4]. Despite this
progress, building a state-of-the-art ASR system remains a
complicated, and expertise-intensive task, especially for less-
resourced and morphologically rich languages.
In ASR systems, usually, phone-based acoustic models
(AMs) and word-based lexical and language models are used.
However, phone-based AMs are not efficient in modeling long-
term temporal dependencies and using words in lexical and
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Andreas Nürnberger
Faculty of Computer Science, Data and
Knowledge Engineering Group,
Otto-von-Guericke University of
Magdeburg
Magdeburg, Germany
andreas.nuernberger@ovgu.de
language models leads to out-of-vocabulary (OOV) problem
and high language model (LM) perplexities, which is a signif-
icant challenge for morphologically rich languages. Amharic
is a morphologically rich language with a high morpheme per
word ratio. Hence, it is challenging to develop a word-based
lexicon and LM. Developing ASR systems for morphologically
rich languages requires selecting a suitable sub-lexical unit to
represent words in the pronunciation dictionary.
To avoid the need for a GMM and to ease the complexity of
the approach, researchers have proposed different techniques,
such as introducing a flat start [5]. However, the complexity of
the approach could not be removed. Since even in the flat start,
the system requires iterative procedures such as generating
forced alignments and decision trees. On the other research
direction, researchers have focused on end-to-end ASR, which
aims to model the mapping between speech and labels directly
with no intermediate components. Graves et al. [6] introduced
the Connectionist Temporal Classification (CTC) objective
function to infer the alignment between speech and label
sequence automatically.
The performance of end-to-end systems is evaluated on
well-known corpora in high-resource languages such as En-
glish and Mandarin. In this work, we investigated the use
of CTC, Convolutional Neural Networks (CNNs), Recurrent
Neural Networks (RNNs) towards an end-to-end ASR system
for the less-resourced and morphologically rich language,
Amharic. We compared the grapheme, phoneme, and sub-
word based end-to-end ASR systems for Amharic. This paper
presents the first results using end-to-end techniques for speech
recognition on less-resourced language Amharic, comparing
sub-word units.
The remainder of the paper is organized as: Section II
describes essential aspects of the Amharic language. Section
III discusses the end-to-end speech recognition systems. In
Section IV, the model architecture is described. Section V
explains the dataset used. Section VI presents the experiments
on the corpus, and the results are discussed under section
VII. The last section, Section VIII, provides conclusions and
highlights of the future work.
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 II. The Amharic Language
Amharic is the official language spoken in Ethiopia. There
are over 25.6 million users, according to Ethnologue1 . Its
orthography represents a consonant-vowel sequence. The lan-
guage has seven vowels and 33 primary characters with each
having seven forms for each consonant-vowel combination (33
x 7 = 231) with additional characters there are 274 distinct
graphemes. The primary difficulties in solving Amharic speech
recognition are the limited availability of corpora and the
complex morphological nature of the language. There are
efforts made to develop both morpheme [7] and syllable-based
[4], [8] ASR systems to overcome language-specific issues.
A. Amharic Morphology
The morphological richness of a language increases the size
of the lexicon in speech recognition. Since Amharic is mor-
phologically rich, the same concept applies to the language.
A single Amharic word could give a large number of mor-
phologically inflected different words. For example, the root
word ጅምር /dʒɨmɨr/ meaning ”started” could provide inflected
words such as ጀመረች/dʒɛmɛrɛtʃ "she started"/, አስጀመረች
/ʔsɨdʒɛmɛrɛtʃ "she made it start"/, አስጀመሩ /ʔsɨdʒɛmɛruɲ "they
made it start"/, ጀመርኩኝ /dʒɛmɛrɨkuɲ "I start"/, ጅማሬ /dʒɨmäre
"beginning"/, ሲጀመር /sidʒɛmɛr "When it started"/, etc. Using
morphemes as a sub-word unit for the Amharic speech recog-
nition system is shown on [7]. However, out of vocabulary
morphemes problem persists. Besides, it is practically difficult
to use the Amharic rule-based morphological analyzer like
HornMorpho [9] for speech recognition purposes. Therefore,
usually, researchers use a data-driven approach for automati-
cally segmenting words into morphemes. Such methods still
require a large number of carefully designed training data.
Therefore, in this paper, we focus on syllables as a modeling
unit.
B. Amharic Syllabification
Syllabification is a process of segmenting words into sylla-
bles. A syllable is a unit of sound composed of a central peak
of sonority, usually a vowel (V), and the consonants (C), which
cluster around this central peak. A syllable can be defined by a
sequence of grammars such as the consonant-vowel-consonant
(CVC) sequence.
Amharic is a syllabic language in which every grapheme
characterizes consonant-vowel assimilation. However, not all
syllables in Amharic satisfy the CV sequence represented
by the graphemes. Amharic syllables can be (C)V(C)(C)
[10], including possible consonant clusters and gemination.
Amharic orthography does not show epenthetic vowel and
geminated consonants that make it challenging to perform
syllabification simply following the templates. For instance,
the word ትምህርት /tmhrt/ meaning ”education” does not show
any vowel when transliterated to IPA. However, the acoustic
evidence shows there is an epenthetic vowel /ɨ/ (see Fig. 1),
/tɨmɨhɨrt/. Gemination is a typical characteristic of Amharic
1 https://www.ethnologue.com/language/amh
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Fig. 1. A waveform and spectrogram for Amharic word ትምህርት /tɨmɨhɨrt/
language which affects the semantics and the syllabification of
a given word. For instance, the word ይመታል /jɨmɛtäll/ mean-
ing ”he hits” its semantics could change when geminated as
ይመታል /jɨmmɛttäll/ meaning ”he will be hit”. The automatic
handling of gemination is not well addressed for the Amharic
language.
A novel syllabification algorithm for Amharic has been
presented in [11]. Acoustic evidence, the maximum onset, and
sonority hierarchy principles have been used to develop a rule-
based syllabification algorithm. All six Amharic syllable tem-
plates are also considered. The algorithm considers gemination
and the unpredictable nature of the Amharic epenthesis vowel.
For our experiment, the algorithm has been re-implemented
in Python with minor improvements. The algorithm is used to
prepare a syllable-based text corpus for the intended experi-
ments after testing the algorithm.
III. End-to-End Speech Recognition
The key idea behind end-to-end systems is to skip the
need for precise alignments between the acoustic and target
state label sequences. It covers directly learning to map
acoustics to word sequences [12]. There are different DNN
layer architectures, which are used in ASR, such as RNNs,
CNNs, and different types of RNNs such as Long Short
Term Memory (LSTM) [12], and Gated Recurrent Unit (GRU)
[13]. LSTM/GRU is useful in modeling sequence-to-sequence
Natural Language Processing problems. They can use their
internal memory to process arbitrary sequences of inputs. This
makes them applicable to tasks such as ASR.
Therefore, two fundamental methodologies have been uti-
lized, both of which exploit LSTM/GRU potential capacity to
remember data for an extended period and follow up on it.
The first of these methodologies utilize LSTM/GRU cells in
the CTC structure to anticipate letter (grapheme) as opposed to
phonetic output [1], [2]. The other methodology uses attention
mechanisms, such as Listen, Attend, and Spell (LAS), which
provides a weighted sum of the hidden activations in a learned
encoding of the input frames as an additional input to the RNN
at each time frame [14], [15].
CNNs used to encode the information from the Mel Fre-
quency Cepstral Coefficients (MFCCs) or the spectrogram
[1], [16]. Then, the encoded data is passed to another con-
volutional layer or LSTM/GRU layer [1]. It finally gives the
Softmax probabilities of each grapheme or phoneme that can
be placed in the transcription. Alternatively, CNN encodes the
information from input features and inputs that data to the
RNN based decoder. It decodes the data and then estimates
the probability of a phoneme or a grapheme by processing
the speech signal. The rectified linear unit (ReLu) is the most
widely used nonlinearity function in training ASR models.
The joint training of the feature stage and the classifier stage
is performed using the back-propagation algorithm. Once the
prediction matrix is gained, the decoding can be done using a
CTC decoding algorithm [6].
To our knowledge, there is no published work so far ap-
plying the end-to-end ASR approach to the publicly available
Amharic speech corpus. The major challenge in building a
speech recognition system for less-resourced languages like
Amharic is the unavailability of well-prepared datasets and
language-specific issues such as the phonetic nature and
orthography of the language. To reduce the need for more
datasets, researchers have studied the use of a language-
adaptive [17], and language-universal techniques [20]. For this
research, we have used approximately 52 hours of Amharic
speech corpus, which is part of our previous work [4].
Fig. 2. CNN+GRU+CTC based ASR Model Architecture.
IV. Model Architectures
The model architectures used in this paper are based on TABLE I
[1], and [2]. However, we focused on keeping the number Comparison of LSTM and GRU Model Parameters.
of parameters minimal and come up with a better result.
Number of Layers Trainable Parameters
In the first model, one layer of 1D CNN in the early 5 GRU ≈1.8 M
stage and five-layered unidirectional GRU followed by batch 5 LSTM ≈2.2 M
normalization (BN) [18] in each layer constitute the first 7 GRU ≈2.6 M
7 LSTM ≈3.6 M
network (see Fig. 2). The prediction from the GRU layers is
feed into the connected layer, which uses softmax computing
to calculate a probability distribution matrix. The network
is trained to receive 13-dimensional MFCCs and generate update gate. represents the Hadamard product (component-
Amharic graphemes, phonemes, or syllables [11]. GRU is wise matrix multiplication). W is the recurrent connection at
selected since experiments showed that the GRU and LSTM the previous hidden layer and current hidden layer, U is the
reach similar accuracy for the same number of layers. However, weight matrix connecting the inputs to the current hidden layer.
the number of trainable parameters is higher for LSTM than Intuitively, the reset gate determines how to combine the new
GRU, given the same number of layers and hidden units. input with the previous memory, and the update gate defines
As it is shown in Table I, there are more trainable parameters how much of the previous memory to keep.
for the LSTM with only 250 hidden units per layer, which A single utterance xi and label yi are sampled from a
makes the training computationally expensive. GRUs are faster training set X = (x1 , y1 ), (x2 , y2 ), …,. Each utterance, xi ,
to train and less likely to diverge as it is also shown in is a time-series of length Ti where every time-slice is a vector
[1]. In this work, we have used the standard GRU with a of audio features, xi t, t = 0, …, ti−1 . The GRU network tries
ReLu activation function (Equation (4)). In standard GRU, the to convert an input sequence xi into a final transcription yi .
hidden state ht is computed as the equation given in Equations The outputs of the network are the Amharic graphemes or
1-5 [12]. phonemes. At each output time-step t, the network makes a
prediction over characters, p(ct |x), where ct is a character in
zt = σ(xt Wz + ht−1 Uz + bz ) (1) the alphabet or the blank symbol. A blank is used to denote
word or syllable boundaries. The model is trained using the
rt = σ(xt Wr + ht−1 Ur + br ) (2) CTC loss function given an input-output pair (x, y) and the
ȟt = tanh(xt W + (rt ht−1 )U + b) (3) current parameters of the network θ. The loss function is
computed as loss(x, y; θ)) and its derivative with respect to the
ȟt = relu(xt W + (rt ht−1 )U + b) (4) parameters of the network 5 θ loss(x, y; θ). This derivative
is then used to update the network parameters through the
ht = (1 − zt ) ht−1 + zt ȟt (5)
backpropagation through time algorithm.
Where matrices Wz , Wr , Uz , Ur and vectors bz , br , b are model In the second model, we similarly kept the network parame-
parameters. σ is sigmoid function r is a reset gate, and z is an ters of the first model. However, we have changed the filter size

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(256) and the hidden unit size (512). Also, we have included
the attention layer after the last GRU layer. The unidirectional
GRU cells are replaced with the bidirectional GRU (BiGRU)
expecting an improvement in the prediction. Attention mech-
anism is combined in the BiGRU allowing it to focus on
certain parts of the MFCCs input sequence while predicting
certain parts of the character sequence. This approach has
proven easier learning and better performance [19]. Attention
mechanism is supposed to improve performance in many tasks
when integrated with RNN networks [15].
In our third model, we have designed 1D deep CNN model
architecture, which gets rid of GRUs in the previous models.
RNNs require more training time and computational resources
than CNNs and Attentions. CNNs with a dropout regular-
ization technique can outperform both RNNs and Attention
mechanisms. In this model, we have used a 10-layer 1D CNN
layers with a filters size of 200 and kernel size 11 without
max-pooling. The dropout layer [20] is used after each CNN
layer with varying dropout rates.
V. Data set
There were only 20 hours of read-speech data available
[21] for Amharic, which makes it challenging to develop
a better speech recognizer system. Collecting a substantial
speech corpus for developing speech recognizer is a costly and
labor intensive task. In our recent study [4], we have prepared
approximately 90 hours of speech corpus, which includes
read-speech, audiobooks, and multi-genre radio shows from
publicly available archives. The dataset is provided for ASR
related extensive research [4]. In this study, we demonstrate
the possibility of having an end-to-end ASR system with a
relatively small subset of the dataset selected.
The subset of the dataset contains a total of 52 hours of
speech data. The dataset is partitioned into training, devel-
opment, and test set. 56% of the training set is augmented
to expand and balance the training set. The dataset partition
contains 16,976 sentences in the training set, 359 in the
development set, and 2780 sentences in the test set. Phoneme
and syllable based labels for each audio is prepared in addition
to the graphemes based labels.
VI. Experiments
The experiments were carried out on the 52 hours of
the Amharic dataset mentioned in Section V. The dataset
contains approximately 20k utterances under the training set
and 3k utterances under the development and test set. In this
experiment, we have used a subset of the dataset published on
[4]. The acoustic features extracted for the experiment consist
of the 13-dimensional MFCCs, with their first and second-
order derivatives. A window size of 20ms with an overlap of
10ms has been used in the estimation of the MFCCs.
All three models mention in Section IV are selected after
testing several model architectures and hyper-parameter tuning.
In the first model, at the early stage 1D CNN layer with a filter
size of 200, a kernel size of 11 and a stride size of 2 is used.
Five stacked GRUs network layers with 250 units in each layer
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are used. The prediction of the final GRU layer fed to the
fully connected layer to calculate and output the probability
distribution matrix in the softmax layer. The output matrix of
the network and the corresponding ground-truth transcriptions
text is input to the CTC loss function. A beam search with
a beamwidth of 100 is used in the CTC decoding algorithm.
In the second model, the attention layer is included after the
last BiGRU layer. The third model is the 1D CNN already
discussed in Section IV. This model is much faster in all the
training. Adam optimizer with a learning rate of 0.0001, beta1
value of 0.9, beta2 value of 0.999, clipnorm value of 1, and
clipvalue of 0.5 is used in all sections of the training. The
experiments were conducted on Geforce RTX 2080 with 8GB
VRAM, 32 GB RAM, and a 16 core processor workstation.
VII. Results and Discussions
As a baseline, we have trained the grapheme-based end-
to-end ASR system for all the three models. We evaluated
the mapping of Amharic graphemes from the MFCCs input.
The first model result showed a character error rate (CER)
of 28.21% on the development set and 32.45% CER on the
test set as it is shown in Table II. Similarly, we have trained
a network, which predicts phonemes to evaluate phoneme se-
quence prediction. We have also trained syllable-based models.
In all the model training, we kept the configuration of the
network identical. The lowest CER achieved is 19.21%, which
is obtained after training 180 epochs on the test set. The result
was collected in the third model which is the syllable-based 1D
CNN model. Phoneme-based models have also shown better
performance compared with the grapheme-based models.
In all experiments, adding layer demands more computing
power and training time to converge with no improvement in
the decoding result. Also, we have increased the number of
neurons in each layer, but that did not improve the result
either. Instead, it took more time to converge. The CNN
model took much less time to train and converge compared
with the other two models. The syllable error rate achieved
in the deep 1D CNN model is much lower (39.98 SER)
compared with the other two models. The result indicates that
1D CNN models can perform better with fewer computation
resources and training time. Adding the attention layer in the
CNN+GRU+CTC model did not improve the CER in any piece
of the training. The deep 1D CNN model could provide a better
result if it is trained for longer epochs since it does not show
over fitting in all the model training.
Compared with the grapheme and phoneme based models,
the syllable based models have shown better CER. This
suggests that syllables are a good modeling unit for Amharic
speech recognition. Besides, for tasks such as spoken content
retrieval(SCR) syllables could be an indexing unit that makes
them a good choice of modeling unit. The result achieved
could be feasible in real-world applications such as spoken
content retrieval since it was shown that an error rate of up to
50% still provided a reasonable SCR accuracy [22].
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