Artificial Intelligence: A Historical Perspective

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The intellectual roots of Artificial Intelligence, and the concept of intelligent machines may be found in Greek mythology. Intelligent artifacts has been appeared in literature since then, with real (and fraudulent) mechanical devices actually demonstrated to behave with some degree of intelligence. Most of these conceptual achievements are listed below under ‘Ancient History.’

After modern computers became available, following World War II, it has become more possible to create programs that perform difficult intellectual tasks. From these programs, general tools are constructed which have applications in a wide spread variety of everday problems. Some of these computational milestones are listed below under ‘Modern History.’

The beginnings of modern AI can be dated to classical philosophers’ attempts to describe human thinking as a symbolic system. But the field of AI wasn’t formally founded until 1956, at a conference at Dartmouth College, in Hanover, New Hampshire, where the term ‘artificial intelligence’ was coined and the revolution has begun.

Technology is growing at a fast pace with the unveiling of the family friendly robots that play the role of a personal assistant at home. Man has already started interacting with computers and smartphones. It is anticipated that social robots shall replace these computers and smartphones in the near future. This work is delving into the design of one such social robot, which supports the above proposition. This project demonstrates an interactive personal assistant robot that is developed using the Raspberry Pi computing engine and Arduino uno board. With the advancement in the technology today, it isn’t a wonder if people get used to having a family friendly personal assistant or social robots in the near future. Many elderly folks and physically disabled people at home require the support of some person for their everyday chores. Sometimes, after a long tiring day at work, people would prefer music and spending some time on the social media for relief from stress. Cooking alone by e-learning may every so often be monotonous and there would arise a need for an interactive environment. In such cases social robots may be very handy. This project proposes the model of one such voice sustained personal assistant robot. The design of the robot is described in a very simple manner and the various aspects of its design are explored.

The main components used for the success of this project are:

  • Rasberry Pi 3 Computer
  • Speaker
  • Motors
  • Ultrasonic Sensors
  • Webcam with Microphone
  • LEDS
  • Battery

The first part of the robot is its body, wheels and a sensor to detect any objects before it. We are interfacing all of these to Raspberry pi . In this project we are placing wheels to the robot, so that it can move on commands and sensors to detect obstacles before it. The second part of the robot is the voice assistant. The voice assistant utilizes the API services. The principle behind this part is that the voice signal from the human is acquired by the web-camera consisting of an in-built microphone. Next, these signals are converted to a format understandable by the computer. The technology used in this conversion is the Text To Speech (TTS) conversion service. Using this converted signal, the computer searches for the appropriate actions to be performed or response to be replied back for the commands given by the human, on the google cloud platform. Once the search has been found the result has to be output on the speaker so as to be heard by the human. This process of conversion of a digital/text signal to a voice signal is implemented using the Google Speech To Text (GSTT) service. This whole process is done on Raspberry pi module.

The robot can be employed with servo motors which can substitute the role of robot arms. These arms could be voice controlled and the robot finds itself useful in some pick and place applications. Instead of manoeuvring the robot using Bluetooth, it could be controlled using voice commands or gesture controls. These features add more essence to the interactive environment created by the robot. Some Multiple Face recognition and synthesis algorithms could be deployed in the robot’s computing engine so that the robot could substitute a security guard/receptionist in the front desk of an organisation. Such robots can be used as home tutors as well as teachers at a primary school to aid the physically challenged children. With the increase in the use of family friendly robots this voice based personal assistant robot finds useful applications in modern homes.

(Components Description Chapter 2)

Raspberry Pi 3

The Raspberry pi is a single computer board with credit card size, that can be used for many tasks that your computer does, like games, spreadsheets and also to play HD video and word processing. It was first established by the Raspberry pi foundation from the UK. It has been ready for public consumption since 2012 with the idea of making a low-cost and effective educational microcomputer for students and children. The sole purpose of designing the raspberry pi board is, to encourage learning, experimentation and innovation for school level students. The raspberry pi board may be a transportable and low price. Maximum of the raspberry pi computers is being used in mobile phones. In the 21st century, the growth of mobile computing technologies is very high, and a huge segment of this being driven by the mobile industries. The 98% of the mobile phones are using ARM technology.

The raspberry pi comes in 2 models, they’re model A and model B. The main difference between model A and B is USB port. Model A board consume less power and that does not include an Ethernet port. But, the model B board includes an Ethernet port and was designed in china. The raspberry pi comes with a collection of open supply technologies, i.e. communication and multimedia system net technologies.In the year 2014, the muse of the raspberry pi board launched the pc module, that packages a model B raspberry pi board into module for use as a locality of embedded systems, to encourage their use.

The Raspberry Pi 3 features are 1.2 GHz quad-core 64-bit Arm Cortex A53 processor, Chip antenna, 4 USB ports, an Ethernet Port, a GPIO, HDMI, 3.5mm Audio Output, WIFI chip, 1GB LPDDR2 for RAM Memory, and a MicroSD slot. The MicroSD card contains the Pi3’s software system and it may also be used for file storage.

The Pi can run the official Raspbian OS, Ubuntu Mate, Snappy Ubuntu Core, the Kodi-based media centers OSMC and LibreElec, the non-Linux based Risc OS (one for fans of 1990s Acorn computers). It may also run Windows ten IoT Core, which is very different to the desktop version of Windows, as mentioned below.

The chip utilized in the Raspberry Pi is reminiscent of a chip utilized in a wireless telephone, and does not become hot enough to need any special cooling. No, it does not need a cooling system. If the processor gets too hot (>85C) it’ll throttle back the speed.

The Foundation’s first 64-bit computing board that also comes with WiFi and Bluetooth built in for the same $35/£30 price. However, the inclusion of integrated Bluetooth four.1 and 802.11n WiFi will please many, as it’ll reduce the need to scour component sites for cheap USB dongles

The raspberry pi board contains a program memory (RAM), processor and graphics chip, CPU, GPU, Ethernet port, GPIO pins, Xbee socket, UART, power source connector. And various interfaces for other external devices. It additionally needs mass storage, for that we use an SD flash memory card. So that raspberry pi board will boot from this SD card similarly as a PC boots up into windows from its hard disk.

Essential hardware specifications of raspberry pi board primarily embrace SD card containing UNIX operating system OS, US keyboard, monitor, power supply and video cable. Optional hardware specifications include USB mouse, powered USB hub, case, internet connection, the Model A or B: USB WiFi adaptor is used and internet connection to Model B is LAN cable.

Newer computers and game consoles have replaced the previous machines wherever most folks learned to program. The creators capitalized on the powerful and cheap processors designed for the booming mobile device arena, and they were able to create an economical, programmable computer with attractive graphics that could boot into the programming surroundings and not break the bank .

The Raspberry Pi may be a mastercard sized single-board pc with associate ASCII text file platform that incorporates a thriving community of its own, almost like that of the Arduino. It can be used in various types of projects from beginners learning how to code to hobbyists designing home automation systems .

There are a few versions of the Raspberry Pi, but the latest version, has improved upon its predecessor in terms of both form and functionality. The Raspberry Pi Model B features:

  • More GPIO
  • More USB
  • Micro SD
  • Lower Power Consumption
  • Better Audio
  • Neater Form Factor

GPIO Pin Out Diagram[5]

This higher-spec variant will increase the Raspberry pi GPIO pin count from twenty six to forty pins. There square measure currently four USB a pair of.0 ports compared to 2 on the Model B. The South Dakota card slot has been replaced with a a lot of fashionable push-push kind small South Dakota slot. It consumes slightly less power, provides better audio quality and has a cleaner form factor. .

To get started you would like a Raspberry Pi three Model B, a 5V USB power offer of a minimum of a pair of amps with a small USB cable, any standard USB keyboard and mouse, associate degree HDMI cable and monitor/TV for show, and a small South Dakota card with the software pre-installed. The NOOBS (New Out Of the Box Software) OS is usually recommended for beginners, and you will select one in all many from the transfer page.

Ultra Sonic Sensors :

As the name indicates,ultrasonic sensors measure distance by using ultrasonic waves.The sensors head emits an ultrasonic wave and receives the wave reflected back from the target.Ultrasonic Sensors measure the distance to the target by measuring the time between the emission and reception[6]. .[image: Outline and detection principle]

Ultra Sonic Sensor[6]

An optical sensing element contains a transmitter and receiver, whereas associate degree inaudible sensing {element|device} uses one inaudible element for each emission and reception. In a reflective model inaudible sensing element, one generator emits and receives inaudible waves alternately. This enables miniaturization of the sensor head.

Distance Calculation:

The distance will be calculated with the subsequent formula:

Distance L = 1/2 × T × C where L is that the distance, T is that the time between the emission and reception, and C is the sonic speed. (The worth is increased by 1/2 as a result of T is that the time for go-and-return distance.)

Features :

  • The following list shows typical characteristics enabled by the detection system.
  • Transparent object detectable
  • Since inaudible waves will mirror off a glass or liquid surface and come to the sensing element head, even clear targets will be detected.
  • Resistant to mist and dirt
  • Detection isn’t suffering from accumulation of dirt or dirt
  • Complex shaped objects detectable
  • Presence detection is stable even for targets like mesh trays or springs.


Speakers area unit one amongst the foremost common output devices used with pc systems. Some speakers area unit designed to figure specifically with computers, whereas others will be connected to any kind of system. Regardless of their style, the aim of speakers is to provide audio output which will be detected by the attender.

Speakers area unit transducers that convert magnetic attraction waves into sound waves. The speakers receive audio input from a tool like a laptop or associate degree audio receiver. This input could also be either in analog or digital kind. Analog speakers merely amplify the analog magnetic attraction waves into sound waves. Since sound waves area unit created in analog kind, digital speakers should initial convert the digital input to associate degree analog signal, then generate the sound waves.

The sound created by speakers is outlined by frequency and amplitude. The frequency determines however high or low the pitch of the sound is. For example, a soprano singer’s voice produces high frequency sound waves, whereas a bass or kick drum generates sounds within the low frequency vary. A speaker system’s ability to accurately reproduce sound frequencies could be a sensible indicator of however clear the audio are going to be. Many speakers embrace multiple speaker cones for various frequency ranges, that helps manufacture a lot of correct sounds for every vary. Two-way speakers generally have a speaker system and a mid-range speaker, whereas triangular speakers have a speaker system, mid-range speaker, and subwoofer.

Amplitude, or loudness, is set by the modification in atmospheric pressure created by the speakers’ sound waves. Therefore, once you crank up your speakers, you’re really increasing the atmospheric pressure of the sound waves they manufacture. Since the signal created by some audio sources isn’t terribly high (like a computer’s sound card), it should ought to be amplified by the speakers. Therefore, most external laptop speakers area unit amplified, which means they use electricity to amplify the signal. Speakers that may amplify the sound input area unit usually referred to as active speakers. You can typically tell if a speaker is active if it’s a volume management or may be obstructed into associate degree outlet. Speakers that do not have any internal amplification area unit referred to as passive speakers. Since these speakers do not amplify the audio signal, they need a high level of audio input, which can be created by associate degree amplifier.

Speakers generally are available in pairs, that permits them to supply stereo sound. This means the left and right speakers transmit audio on 2 fully separate channels. By exploitation 2 speakers, music sounds far more natural since our ears area unit accustomed hearing sounds from the left and right at a similar time. Surround systems may include four to seven speakers (plus a subwoofer), which creates an even more realistic experience[7].

Web cam with microphone

A electrical device could be a transducer that converts sound into associate degree electrical signal.

Microphones area unit utilized in several applications like telephones, hearing aids, public address systems for concert halls and public events, picture production, live and recorded audio engineering, recording, megaphones, radio and televisionbroadcasting, and in computers for recording voice, speech recognition, VoIP, and for non-acoustic purposes such as ultrasonic sensors or knock sensors[8].

Webcam with microphone[8]

Several differing kinds of electro-acoustic transducer area unit in use, that use completely different ways to convert the gas pressure variations of a undulation to associate electrical signal. The most common area unit the dynamic electro-acoustic transducer, that uses a coil of wire suspended in an exceedingly magnetic field; the microphone, that uses the vibratory diaphragm as a capacitance plate, and also the electricity electro-acoustic transducer, that uses a crystal of piezoelectricmaterial. Microphones generally have to be compelled to be connected to a preamplifier before the signal is recorded or reproduced.


An electric motor is associate electrical machine that converts electricity into energy.

The motor used for this project has 150 RPM speed, which is used for the movement of the bot in the direction commanded by us.s]

Data Mining

Cloud Speech to Text conversion

Human move with one another in many ways that like countenance, eye contact, gesture, primarily speech. The speech is primary mode of communication among individual and conjointly the foremost natural and economical variety of exchanging data among human in speech [1]. Speech-to-text conversion (STT) system is wide employed in several application areas. In the instructional field, STT or speech recognition system is that the best on deaf or dumb students. The recognition of speech is one the foremost challenges in speech process. Speech Recognition can be defined as the process of converting speech signal to a sequence of words by means of Algorithm implemented as a computer program [9]. Basically, speech to text conversion (STT) system is distinguished into two types, such as speaker dependent and speaker independent systems [9]. Speech recognition is incredibly quality case once process on arbitrarily variable analogue signal like speech signals. Thus, in speech recognition system, feature extraction is that the main a part of the system.

There are various methods of feature extractions. In recent researches, several feature extraction techniques square measure unremarkably used like Principal part Analysis (PCA), Linear Discriminant Analysis (LDA), freelance part Analysis (ICA), Linear Predictive Coding (LPC), Cepstral Analysis and Mel-frequency cepstral (MFCCs), Kernal based feature extraction based approach, Wavelet Transform and spectral subtraction [9]. In this paper, Mel Frequency Cepstral Coefficients (MFCC) methodology is employed. It is supported the characteristics of the human ear’s hearing, which uses a nonlinear frequency unit to simulate the human auditory system. Mel frequency scale is wide wont to extract options of the speech. Mel-frequency cepstral features provide the rate of recognition to be efficient for speech recognition as well as emotion recognition system through speech [9]. Moreover, Vector quantisation (VQ), Artificial Neural Network (ANN), Hidden Markov Model (HMM), Dynamic Time Warping (DTW) and various techniques are used by the researchers in recognition. Among them, HMM recognizer is presently dominant in several applications. Nowadays, STT system is fluently employed in several management systems, mobile phones, computers and so forth. Therefore, speech recognition system is a lot of and a lot of common and helpful in our daily lives. In the system, MFCC and HMM are implemented by MATLAB. Flow chart is shown in fig 3.1.


End Point Detection

Classification of speech into voiced or unvoiced sounds provides a helpful basis for resulting process. A threeway classification into silence/unvoiced/voiced extends the possible range of further processing to tasks such as stop consonant identification and endpoint detection for isolated utterances [9]. In howling atmosphere, speech samples containing unwanted signals and ground noise square measure removed by finish purpose detection methodology. End point detection method is based on the short-term log energy and short-term zero crossing rate [9]. The logarithmic short-term energy and zero crossing rates are calculated using some equations(equations are not included in this project).

Mel Frequency Cepstral Coefficient (MFCC)

Feature extraction is the most important part of the entire system. The aim of feature extraction is to reduce the data size of the speech signal before pattern classification or recognition. The steps of Mel frequency Cepstral Coefficients (MFCCs) calculation are– framing, windowing, Discrete Fourier Transform (DFT), Mel frequency filtering, logarithmic function and Discrete Cosine Transform (DCT).Fig.3.2 shows the block diagram of MFCC process.


It is the first step of the MFCC. It is the method of block of the speech samples obtained from the analogue to digital conversion (ADC) of the word, into the amount of frame signal with 20- 40ms frame time length. Overlapping is needed to avoid loss of information. Windowing: so as to scale back the discontinuities at the beginning and finish of the frame or to be swish of the primary and last points within the frame, windowing function is used. DFT: separate Fourier rework (DFT) is employed because the quick Fourier rework (FFT) algorithmic rule. FFT converts every frame of N samples from the time domain into the frequency domain. The calculation is a lot of precise in frequency domain instead of in time domain.

Mel frequency filtering: The voice signal doesn’t follow the linear scale and therefore the frequency place FFT is thus wide. It is sensory activity scale that helps to simulate the manner human ear works. It corresponds to higher resolution at low frequencies and fewer at high. Logarithmic function: power transformation is applied to absolutely the magnitude of the coefficients obtained once Mel-scale conversion. The absolute magnitude operation discards the section data, making feature extraction less sensitive to speaker dependent variations. DCT: separate trigonometric function rework (DCT) converts the Mel-filtered spectrum into the time domain since the Mel Frequency Cepstral Coefficients area unit used because the time index in recognition stage.

Hidden Markov Model Recognizer

In recognition or classification of the speech signal, there area unit several approaches to acknowledge the check audio file. The methodologies of speech recognition are: ANN, GMM, DTW, HMM, formal logic and varied alternative ways. Among them, HMM techniques area unit wide employed in several applications than the other ones. There area unit four kinds of HMM model employed in speech process. Details of HMM models are given in [9]. The phonemes in speech follow the left to rightsequences, that the structure of HMM may be a left-to-right structure. The states of HMM model represent the word or acoustic phonemes in speech recognition. The number of states of HMM model is every which way chosen to model. The choice of the amount of states causes to vary the feature vectors or observations. It affects the popularity rate or accuracy of speech recognition.The most versatile and productive approach to speech recognition up to now has been Hidden Andrei Markov Models (HMMs). HMM is that the well-liked applied math tool for modeling a good vary of your time series knowledge. In speech recognition area, HMM has been applied with great success to problem such as part of speech classification [9]. HMM word model 𝜆 is composed of initial state probability (𝜋), state transition probability (A) and symbol emission probability (B). In HMMbased speech recognition system, there exist 3 main issues known as analysis, secret writing and learning issues. The training and testing algorithm of HMM are discussed in details [9]. The chance of observations or chance given the model determines the expected recognized word.


The flowchart of speech to text conversion is illustrated in Fig .3.3. To convert input speech to text output, the four main steps are developed by using MATLAB.These steps are speech database, preprocessing, feature extraction and recognition. Firstly, 5 audio files area unit recorded with the assistance of pc. Each audio file contains 10 totally different pronunciation audio files. So, there area unit total of fifty audio files area unit recorded in speech information. The speech signals at low frequencies have a lot of energy than at high frequencies. Therefore, the energies of signal area unit necessary to be boost at high frequencies. According to the saturation of setting, the unwanted noise may affect the recognition rate worse. This downside are often overcome by finish purpose detection technique. After preprocessing stage is finished, the speech samples are extracted to features or coefficients by the use of Mel Frequency Cepstral Coefficient (MFCC). Finally, these MFCC coefficients area unit used because the input of Hidden Andrei Markov Model (HMM) recognizer to classify the specified word. The desired text output can be generated by HMM method even if the test audio file is included in the existing replaced by their corresponding whole words or phrases, and so on.

Cloud text to speech conversion

Google Cloud Text-To-Speech allows developers to synthesize natural sounding speech with thirty voices, accessible in multiple languages and variants. It applies DeepMind’s ground breaking research in WaveNet and Google’s powerful neural networks to deliver high fidelity audio.With this easy-to-use API, you can create lifelike interactions with your users, across many applications and devices

By speech synthesis we are able to, in theory, mean any kind of synthetization of speech. For example, it can be the process in which a speech decoder generates the speech signal based on the parameters it has received through the transmission line, or it can be a procedure performed by a laptop to estimate some quite a presentation of the speech signal given a text input. Since there’s a special course concerning the codecs (Puheen koodaus, Speech Coding), this chapter will concentrate on text-to-speech synthesis, or shortly TTS, which will be often referred to as speech synthesis to simplify the notation. Anyway, it’s sensible to stay in mind that regardless of what quite synthesis we tend to area unit managing, there are similar criteria in regard to the speech quality. We will come to the present topic when a short TTS motivation, and the rest of this chapter will be dedicated to the implementation point of view in TTS systems. Text-to-speech synthesis could be a analysis field that has received tons of attention and resources throughout the last number of decades – for glorious reasons. One of the foremost fascinating concepts (rather futurist, though) is the fact that a workable TTS system, combined with a workable speech recognition device, would actually be an extremely efficient method for speech coding (Huang, Acero, Hon, 2001).

It would give extraordinary compression magnitude relation and versatile prospects to settle on the sort of speech (e.g., breathless or hoarse), the fundamental frequency along with its range, the rhythm of speech, and several other effects. Furthermore, if the content of a message needs to be changed, it is much easier to retype the text than to record the signal again. Unfortunately this sort of a system doesn’t nonetheless exist for big vocabularies. Of course there {are also|also area unit|are} various speech synthesis applications that are nearer to being accessible than the one mentioned on top of. For instance, a phonephone inquiry system wherever the data is often updated, will use TTS to deliver answers to the purchasers. Speech synthesizers also are vital to the visually impaired and to those that have lost their ability to talk. Several different examples are often found in existence, like being attentive to the messages and news rather than reading them, and using hands-free functions through a voice interface in a car, and so on.

Implementation of TTS

The process of reworking text into speech contains coarsely 2 phases: initial the text goes through analysis and so the ensuing data is employed to come up with the speech signal. In the block diagram shown in Figure , the former phase actually contains not only text analysis but also phonetic analysis in which the graphemes are converted into phonemes. The generation of the speech signal also can be divided into 2 sub-phases: the search of speech segments from a info, or the creation of these segments, and the implementation of the prosodic features. These phases will be further discussed in the following. Text analysis is all concerning reworking the input text into a ’speakable’ kind. At the minimum, this contains the normalization of the text so that numbers and symbols become words, abbreviations are replaced by their corresponding whole words or phrases, and so on.

This method generally employs an oversized set of rules that attempt to take some language-dependent and context-dependent factors into consideration. The most difficult task within the text analysis block is that the linguistic analysis which suggests syntactical and linguistics analysis and aims at understanding the content of the text. Of course, a computer cannot understand the text as humans do, butstatistical methods are used to find the most probable meaning of the utterances. This is necessary as a result of the pronunciation of a word could depend upon its that means and on the context (for instance, the word record is pronounced in different ways depending on whether or not it’s a verb or a noun). Finally, the text analysis block is meant to produce manner of speaking info to the next stages. It can, as an example, signify the positions of pauses based on the punctuation marks, and distinguish interrogative clauses from statements so that the intonation can be adjusted accordingly.

Phonetic analysis converts the ortographical symbols into descriptive linguistics ones employing a sound alphabet. We have already seen on this course IPA, the alphabet of the International Phonetic Association. IPA contains not solely sound symbols however conjointly diacritic marks and alternative symbols associated with pronunciation. Since the IPA symbols square measure rather sophisticated and there square measure many symbols that can’t be found in typewriters, other phonetic alphabets have also been developed. They are higher compatible with computers and infrequently supported American Standard Code for Information Interchange characters. Examples of such alphabets square measure SAMPA (Speech Assessment ways – Phonetic Alphabet), Worldbet and Arpabet. However, there’s no usually accepted, common phonetic alphabet and therefore separate speech synthesizers often use their own special alphabets (Lemmetty, 1999). The degree of challenge in phonetic analysis is powerfully language dependent – Finnish is truly one in every of the simplest languages during this respect as a result of the pronunciation isn’t thus different from the written form of the utterance.

Prosody may be a conception that contains the rhythm of speech, stress patterns and intonation. The attachment of sure manner of speaking options to artificial speech employs a collection of rules that square measure supported the manner of speaking analysis of natural speech. Prosody plays a really necessary role within the quality of speech and moreover, manner of speaking options carry innumerable info regarding the speaker, for instance his or her emotional state, and even the social background. In follow, generating natural sounding prosody in massive vocabulary speech synthesis continues to be a remote goal as a result of the modeling of prosody is such a problematic task. Some hierarchical rules are developed to regulate the temporal order and harmonic, and this has created the flow of speech in synthesis systems somewhat a lot of natural sounding.

Speech synthesis block finally generates the speech signal. This can be done either supported a constant quantity illustration, during which case sound realizations square measure created by machine, or by choosing speech units from a information. In the latter methodology, a complicated search method is performed so as to seek out the acceptable sound, diphone, triphone, or other unit at each time. Whichever methodology is chosen, the resulting short units of speech are joined together to produce the final speech signal. One of the most important challenges within the synthesis stage is truly to create positive that the units hook up with one another in a very continuous method in order that the quantity of audible distortion is minimized.

The following subsections describe the most principles of the 3 most typically used speech synthesis methods: formant synthesis, concatenative synthesis, and articulatory synthesis. More information about this subject can be found, for example, in the Master’s Thesis of Sami Lemmetty (see the literature list at the end of this chapter).

Formant Synthesis

This is the oldest methodology for speech synthesis, and it dominated the synthesis implementations for a long time. Nowadays the concatenative synthesis is additionally a really typical approach. Formant synthesis is predicated on the well-known supply-filter model which suggests that the concept is to come up with periodic and non-periodic source signals and to feed them through a resonator circuit – or a filter – that models the vocal tract.

The principles square measure therefore terribly straightforward, that makes formant synthesis versatile and comparatively simple to implement. In distinction to the ways delineate below, formant synthesis are often accustomed turn out any sounds. On the opposite hand, the simplifications created within the modeling of the supply signal and vocal tract inevitably cause somewhat unnatural sounding result.

In a discourteously simplified implementation, the supply signal are often AN impulse train or a serration wave, along with a random noise element. To improve the speech quality and to realize higher management of the signal, it is naturally advisable to use as accurate model as possible. Typically the adjustable parameters embody a minimum of the elemental frequency, the relative intensities of the voiced and unvoiced supply signals, and therefore the degree of adjustment. The vocal tract model typically describes every formant by a try of filter poles in order that each the frequency and therefore the information measure of the formant are often determined. To make ingelligible speech, at least three lowest formants should be taken into account, but including more formants usually improves the speech quality. The parameters controlling the 56 frequency response of the vocal tract filter – and those controlling the source signal – are updated at each phoneme. The vocal tract model are often enforced by connecting the resonators either in cascade or parallel kind. Both have their own blessings and shortcomings however they’re going to not be mentioned here. In addition to the resonators that model the formants, the synthesizer will contain filters that model the form of the vocal organ undulation and therefore the lip radiation, ANd conjointly an anti-resonator to better model the nasalized sounds.

Concatenative Synthesis

This is the thus known as cut and paste synthesis during which short segments of speech ar elite from a pre-recorded information and joined one when another to supply the required utterances. In theory, the employment of real speech because the basis of artificial speech brings regarding the potential for terribly prime quality, however in apply there ar serious limitations, mainly due to the memory capability needed by such a system. The longer the chosen units ar, the fewer problematic concatenation points will occur in the synthetic speech, but at the same time the memory requirements increase. Another limitation in concatenative synthesis is that the robust dependency of the output speech on the chosen information. For example, the personality or the affective tone of the speech is hardly controllable. Despite the somewhat plain nature, concatenative synthesis is well suited for certain limited applications. What is the length of the selected units then? The most common decisions ar phonemes and diphones as a result of they’re short enough to achieve enough flexibility and to stay the memory necessities cheap. Using longer units, such as syllables or words, is impossible or impractical for several reasons. The use of diphones within the concatenation provides rather sensible potentialities to require account of coarticulation as a result of a diphone contains the transition from one phone to a different and also the latter half the primary phone and also the former half the latter phone.

Consequently, the concatenation points are situated at the middle of every phone, and since this is usually the most steady part of the phoneme, the amount of distortion at the boundaries can be expected to be minimized. While the enough range of various phonemes during a information is often around forty – fifty, the corresponding range of diphones is from 1500 to 2000 however a synthesizer with a information of this size is usually implementable (Lemmetty, 1999). On the opposite hand, the employment of phonemes is that the most versatile method of generating numerous utterances, at least if we ignore the fact that certain phonemes (e.g., plosives) are fairly not possible to break free a speech signal to their own segments. In each phone and diphone concatenation, the greatest challenge is the continuity. To avoid sounding distortions caused by the variations between sequent segments, at least the fundamental frequency and the intensity of the segments must be controllable. The creation of natural prosody in artificial speech is not possible with the current ways however some promising ways for obtaining obviate the discontinuities have naturally been developed. Finally, concatenative speech synthesis is afflicted by the hard method of making the information from that the units are elite. Each phone, together with all of the needed allophones, must be included in the recording, and then all of the needed units must be segmented and labeled to enable the search from the database. Some of these operations is automatized to sure extent.

Articulatory Synthesis

Compared with the opposite synthesis ways given during this chapter, pronunciation synthesis is far and away the foremost difficult in relevance the model structure and machine burden. The idea in articu57 latory synthesis is to model the human speech production mechanisms as perfectly as possible. The implementation of such a system is extremely tough and so it’s not wide in use however. Experiments with pronunciation synthesis systems haven’t been as victorious like alternative synthesis systems however in theory it’s the simplest potential for high-quality artificial speech. For example, it is impossible to use articulatory synthesis for producing sounds that humans cannot produce (due to human physiology). In alternative synthesis ways it’s attainable to provide such sounds, and the problem is that these sounds are usually perceived as undesired side effects. The pronunciation model additionally allows additional correct transient sounds than alternative synthesis techniques.

Articulatory synthesis systems contain physical models of each the human vocal tract and therefore the physiology of the vocal cords. It is common to use a collection of space functions to model the variation of the cross-sectional space of the vocal tract between the vocal organ and therefore the lips. The principle is therefore just like the one that has been seen among the acoustic tube model. The pronunciation model involves an outsized range of management parameters that area unit used for the terribly elaborate adjustment of the position of lips and tongue, the lung pressure, the tension of vocal cords, and so on. The data that’s used because the basis of the modeling is sometimes obtained throug the X-ray analysis of natural speech (Lemmetty, 1999). As expected, such analysis is also very troublesome.


This module works on the keyword of “wiki”. The system asks for what you would like to learn about. Then the request is created to the Wikipedia API for the specified question. It generates the outline of knowledge} concerning the question and therefore the data is output through the mike to the beholder in audio kind. In case of failure, the error message is generated voice communication “unable to succeed in lexicon of wiki”.


News module is dead by victimization the keyword “news”. The headlines of prime articles square measure retrieved from the net victimization Google news. The system tells the user of these headlines and asks the user if any of those articles ought to be sent to the user’s email address. If the user specifies the quantity of the article to be sent, the article is distributed to the required email address. Otherwise, no further action is taken. If any failure happens in retrieving headlines or causation articles, a corresponding error message is generated.


This module tells the user regarding the weather of the placement whose station symbol is laid out in the profile of the user. This module is dead by victimization the keyword “weather”. The weather info is taken from the weather underground service which incorporates the small print of temperature, wind speed and direction etc. It generates a slip-up message, if the information cannot be retrieved for the specified location.


Joke module is used for amusement functions by the user. This module works on the keywords “joke” or “knock knock”. The jokes utilized in this module square measure predefined in a very computer file from that the jokes square measure browse in a very random order. A begin and finish line is gift in each joke to differentiate it from others gift within the file. All the lines of a joke square measure spoken by the system within the mere order solely.

Future Scope

It has been well documented that there’ll be increase within the range of robots over consecutive decade. According to the Boston Consulting cluster, by 2025, robots will perform 25% of all labor tasks. This is because of enhancements in performance and reduction in prices. The u. s., along with Canada, Japan, South Korea, and the United Kingdom, will be leading the way in robot adoption. The four industries leading the charge square measure laptop and electronic products; electrical instrumentality and appliances; transportation equipment; and machinery. They will account for seventy fifth of all robotic installations by 2025.

The remaining segments enclosed robot robots (including assistant/companion robots), telepresence robots, powered human exoskeletons, surgical robots, and autonomous mobile robots. Combined, they were calculable to own had but fifty,000 units put in.

Humanoid robots, while being one of the smallest groups of service robots in the current market, have the greatest potential to become the industrial tool of the future. Companies like Softbank AI have created human-looking robots to be used as medical assistants and teaching aids. Currently, robot robots square measure excelling within the medical business, particularly as companion robots.

Cite this paper

Artificial Intelligence: A Historical Perspective. (2022, Nov 27). Retrieved from https://samploon.com/artificial-intelligence-a-historical-perspective/

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