Speech Recognition Using attention-based sequence-to-sequence methods

Abstract—Speech is one of the most important and prominent manner to communicate among human being. It also has capacity to become a kind of medium when facing the human computer interaction. Speech recognition has become a popular area across research institutes and the Internet-related companies. This paper presents a brief overview on two main steps of speech recognition which are feature extraction and training model using deep learning. In particular, five art-of-state methods using attention-based sequence-to-sequence model for speech recognition training process are discussed.

Keywords-speech recognition; attention mechanism; sequence to sequence; neural transducer; Mel-frequency cepstrum coefficient

I. INTRODUCTION
Natural language refers to a kind of language that evolves naturally with culture, and it is also the primary tool of human communicating and thinking. While speech recognition, as the name suggests that takes natural language speech as the input into the model, and the output is the text of the speech. In other words, it is converting speech signals into text sequences. It is simple for humans to convert speech audio into text manually. Still, when facing large amounts of data, it takes plenty of time, and it is, to some extent, very difficult or impossible to covert in real-time for humans. Moreover, there are hundreds of languages in the world, so that few experts can master multiple languages simultaneously. As a result, people expect that machine learning can help people achieve this task.

At present, the typical steps of speech recognition include preprocessing, feature extraction, training, and recognition. For the feature extraction, because the speech signal is volatile, even if people try hard to say the same two sentences, the signals of which always have some differences. So, feature extraction of speech is difficult for computer scientists.

In this paper, we introduce the main process of speech recognition. For feature extraction, we introduce one of the most popular approaches of it which is called the Mel-frequency Cepstrum Coefficient (MFCC) [1]. For the training part, it is evident that the length of input (speech vectors sequence) and output (text vectors sequence) is probably different. Input is determined by humans (etc., select 25ms speech), and the specific output length is determined by the model itself. Thus, Sequence-To-Sequence (Seq2Seq) based models are most widely used nowadays.

The remainder of this article is organized as follows. In Section II, the MFCC feature extraction approach is illustrated. In Section III, we describe the basic attention mechanism, as well as five training methods based on it: Listen, Attend, and Spell Connectionist Temporal Classification, RNN transducer, Neural Transducer, and Monotonic Chunkwise Attention. Finally, concluding remarks are contained in Section IV.

II. FEATURE EXTRACTION

Because of the instability of speech signals, feature extraction of the speech signal is very difficult. It exists different features between each word. For each word, there are differences among different people, such as adults and children, male and female. Even for the same person and the same word, there also exists changes for a different time[2]. Mel-Fre-Frequency-Doppler is proposed based on the different auditory characteristics of human ears. It uses the nonlinear frequency unit, which names Mel frequency [4], to simulate the human auditory system [8,10,11,17,18]. The calculation method is shown in Formula (1):

Figure 1 shows the construction of the MFCC model.

The original acoustic wave through the window and other pre-processing, then we obtain the frame signal.

Because it is difficult to observe the characteristics of the signal in the time domain, transforming it into the energy distribution in the frequency domain can solve this problem. The energy distribution in the spectrum, which represents the characteristics of different sounds, is obtained by fast Fourier transform.

After the fast Fourier change of the speech signals is completed, Mel frequency filtering is performed [3]. The specific step is to redefine the filter bank composed of triangular bandpass filters, assuming that the center frequency of each filter is , is the low-pass frequency within the coverage range after the cross overlap of three filters, is the high pass frequency. Then the calculation method is shown as

We can obtain the output spectrum energy generated by each filter, and then the data should be transformed into a logarithm. Finally, the discrete cosine transform is converted to the time domain to obtain the final MFCC. In MFCC, the main advantage is tha