ITRM20090089A1 - EQUALIZATION TECHNIQUES OF THE AUDIO SIGNAL IN FIXED AND MOBILE TELEPHONY. - Google Patents

Description

â € œEQUALIZATION TECHNIQUES OF THE AUDIO SIGNAL IN

FIXED AND MOBILE TELEPHONYâ €

DESCRIPTION

Sector to which the invention pertains

The present invention refers to equalization techniques that can be used in telephony. More particularly by means of the present technique, the reproduction of audio frequencies is such that whoever is using the telephone does not hear the interference and disturbance effects usually introduced by conventional microphones / loudspeakers on the user terminal used in mobile telephony. It therefore concerns both micro-tuned electronic devices that can be used as equalizers but also telephone network devices and the related management software in charge of optimizing the transmission of the audio signal.

State of the art

LEGEND GSM: Global system for mobile communications

HLR: Home Location Register is the database used for the permanent storage of subscriber data with the relative profiles of the assigned services.

MSC: Mobile Switching Center has the function of switching voice and data traffic to the GSM network

VLR: Visitor Location Register is a database that contains the information of users who are temporarily assigned to the operator thanks to the roaming functions.

VoIP: Voice over IP, voice signal transmission over IP / data network. During a telephone conversation one of the most important problems deriving from the use of standard reproduction systems such as those provided by default in the mobile phone, is the interference produced by the signals and noises coming from different sound sources. This phenomenon goes by the name of audio intermodulation received by the microphone and the preamplifier so that all the sounds that are mixed flow into the speaker for the excitation, at the same time, of the relative coil.

In particular, the impossibility of filtering, identifying and listening to the single voice among a group of sound sources, the effect of sound interference due to the arrangement of the microphone and the loudspeaker inside the same mobile phone, and the different types and groups of sounds that come for example from the outside or from nearby audio equipment often lead the hearing-impaired user to not have an effective, clean and distinct listening.

The frequent annoying phenomenon of autophony, which derives from impaired listening to one's voice, is also known in the state of the art. This disturbance is attributable to the use of the earphone which, although equalized, still causes an alteration in the perception characteristics of the ear. These problems generally cause difficulties in verbal comprehension, and a reduction in the comfort of use of the devices - the so-called hearing aids - that correct and / or help listeners with limited hearing abilities. The latter can assume, due to the aforementioned limited perception, an attitude of closure with the outside world.

Furthermore, it should be emphasized that the exposure of the ear in proximity to the speakers kept in close contact produces a signal that exceeds 140 db: in fact, raising the volume even operates in a â € œpainâ € band for the ™ listener. With this type of increase in the amplitude of the sound signal, what is still healthy is damaged in the user's hearing capacity, further degrading it. In practice, a positive reaction process is established whereby the more the user hears badly, the more he amplifies the sound level of the incoming signal, the more it damages the ear that listens, the user therefore hears even more badly.

As known, at the state of the art equalizers are electronic devices in charge of selective amplification. Some constituent parts of the equalizer 37 are similar to amplifiers used in audio transceiver systems, even if, more particularly in the case of vocal signal, the amplification of the signal is directed to selectively amplify only some frequencies. Equalizers usually comprise an audio frequency amplifier with several filters at different slopes and frequency responses, hence an adjustable and selectable frequency response. They are usually used to compensate for hearing sensitivity using pre-configured electronic means so as to regulate the optimal hearing level that must be achieved. In more general terms, three types of mobile phone equalizer are known, equalizer:

i- for actual mobile phone (in the sense that it is contained within the same casing as the mobile phone),

ii- integrated in the earphone that is usually used near the ear,

iii- external, distinct from both the mobile phone and the headset configured as a box containing the equalizer to be used as an accessory placed between the headset and the mobile phone.

The person who has hearing loss problems however avoids operating with two amplification systems at the same time in order not to generate interference between the sound reproduction of the mobile phone and the prosthesis; instead he would prefer to listen only from a mobile phone equipped with a suitable equalizer.

On the other hand, at the moment, digital technologies applied to traditional prostheses have given birth to digitally programmable devices with traditional sound processing circuits (microphone, amplifier, receiver) that can be adjusted via an external computer. The audiometric technician is able to adapt the sound processing to the user's needs via computer through specific controls. In fact, digitally programmable techniques are currently known which are well adaptable to the listening needs of the user. In the event of a change in the level of hearing loss, they can, within certain limits, be reprogrammed for new hearing needs, having diversified ranges of application: mild, medium, severe and deep.

After all, digital has introduced a new philosophy of approach to the user with limited hearing ability. Before it was the user who was forced to adapt to the hearing aid, nowadays it is the prosthesis that can adapt to the user's needs. With this type of aid it is possible to intervene on fifteen, twenty even thirty different parameters that define the modality and type of sound processing to be sent to the user's ear.

Summary of the invention

The purpose of the present invention is to realize an equalizer device manageable by the user, possibly directly on the mobile phone display, with which the user himself (in particular with reduced hearing capacities), can listen to sounds at audio frequencies without the disturbances resulting from interference or unwanted noise.

Another object of the present invention is to provide an equalization technique of the audio signal in fixed and mobile telephony, which allows to optimize the listening to users with reduced hearing capacities through an interactive adjustment procedure of the selective filters of the audio frequencies. used and understood in the sound signal related to speech that the two telephones taking part in the conversation exchange.

It is still the object of the invention to provide a solution that features a mobile phone that uses differential microphones having optimized frequency responses and optimized sensitivity characteristics compared to common microphones.

Finally, the aim of the present invention is to provide an equalization technique of the audio signal in fixed and mobile telephony that uses hardware components of the mobile phone and standard techniques in the field of telephony and communications on the cellular telephone network and on the fixed telephone network, to in order to make the system itself cost-effective, highly reliable and easy to use.

Based on these objectives, a programmable and interactive equalizer is provided comprising a differential microphone, dynamically programmable filtering means, an adjustable power amplifier, a sound transducer, all powered by the cell phone battery. The capacitive filtering means are provided with programmable memory and with the relative interface constituted in the form of an integrated circuit which comprises at least one oscillator associated with them. The keypad and the mobile phone display constitute the I / O means that allow the user to enter and view the adjustment parameters.

The power amplifier, on the other hand, corresponds to the terminal section that drives the loudspeaker powered by the same battery 33 as the mobile phone and is also currently produced as an integrated circuit.

A series of solutions are also provided at the level of telephone network architecture to provide an aid to the hearing-impaired user both by arranging an equalization in real time (which, based on the environmental noise conditions, prepares an equalization on particular frequencies), and by providing a visual aid in textual form that translates the vocal signal into a sequence of words in text format that can be interactively displayed on the display 31 of the mobile phone, or of a PC in case of communication with a VoIP technique over TCP / IP.

Brief description of the drawings

The invention will be better understood by reading the following description, provided purely as an example and made with reference to the drawings, in which:

FIGURE 1 is a block diagram showing the network connections between the two telephones that intervene in the conversation for equalization according to the present invention,

FIGURE 2 is a block diagram of the component sections of a mobile phone according to the present invention.

Description of favorite achievements

The invention through the selection menus present on the mobile phone display 31 allows the choice of frequencies appropriate to the hearing abilities of the user of the mobile phone. In fact, as is well known, what allows people with impaired hearing capacities to listen is never the increase in volume (of amplification), but only the best reception of particular â € œtonesâ € € (frequencies) which, when appropriately amplified, intervene on the sections of a non-optimized frequency response curve.

The principle of the present invention is based on the realization of a technique which allows the regulation and filtering in real time of the frequencies, amplified dynamically, through a dedicated menu. This menu is shown on the telephone display 31, provided to the user in graphic or textual mode, so that he can have a series of equalization procedures for the incoming sound signal. More specifically, each component of the frequency spectrum of the sound signal is selected appropriately and the user can intervene on the known parameters following a typically audiometric protocol that can be performed either at the time of activation (preset) of the mobile phone or in time. real every time a communication is established. In the latter case, the obvious advantage is to be able to adapt the equalizer to the specific environmental conditions of the moment which can be more or less noisy, more or less disturbed on particular frequencies instead of on others. In other words, on the one hand, an equalization operation of the single mobile phone can be performed that corresponds to an adjustment procedure between base radio station 2 and telephone A, B. Or the user goes to a distributor shop - specialized center managed by the network operator, with audiometric center functions - so that a configuration / initialization of the mobile phone is carried out

In both cases, a procedure for examining the auditory abilities of the user is provided, which operates through an interactive menu according to the criterion of dichotomous or binary search, according to which, just as when you are looking for a word in a vocabulary , the â € œgoodâ € half is excluded from time to time until the number of pages is reduced to one: either the word is there or it means that it is not there.

In this way, the transposition of what in mathematics goes under the name of interpolated research is used for this type of regulation. In fact, the regulation of the audio signal spectrum, in the present invention, has a typically adaptive characteristic since the detection of noise and disturbances is carried out by sampling on a reduced spectrum of the audio frequencies, a fine regulation being implemented, as already stated, dichotomous type. For example, a frequency of 1000 hertz is sent first and then a frequency of 500 hertz and one asks, as is done by the ophthalmologist, if the first one has been heard better or the second better, if one has listened better the first (1000 Hz) is considered a current interval between 500 and 750 and asks whether the first has been heard better or better the second; if, on the other hand, the second frequency had been heard better, the frequency at 1000 hertz and the one at 750 hertz were sent and he wondered if the first frequency had been heard better or the second better. This procedure takes place iteratively until it is possible to identify which frequency is worst listened to.

As stated, this adjustment activity can be performed with a preset, one-off, at a specialized center or whenever the user requests it according to his own needs, by interfacing with a radio base station 2 and activating an interactive menu from his mobile phone. .

The regulation is based on the voice machines that operate, as in the call to the Call Center, with automatic responders. Therefore, when configuring the mobile phone with equalizer 37, a voice machine is expected to answer and ask you to type various keys 1, 2 or 3 â € ¦ to have specific selection functions of the frequency of the activated bandpass filter and of the amplification relative. Interactively it will be possible to see on the mobile phone screen 31 a display of the frequencies which are activated and deactivated for example by means of a bar graph.

On the basis of the results of this empirical operating mode, of detection, of the frequencies heard with less intensity, the test of the dedicated telephone set according to the present invention provides, below, the verification of the insertion gain, which consists in the measurement of the amplification delivered by the equalizer at the eardrum level.

This is a real speech audiometry procedure with and without competition. As the tests are repeated with and without acoustic equalizer. The ability to understand commonly used words and phrases is evaluated. This latest test also provides the functional gain of the phone.

The obvious advantage is that there is no longer any need to keep the volume of cellular devices high (thus avoiding that others listen in spite of them, what the two interlocutors of the telephone conversation say to each other).

The telephone device used - that is to say the actual mobile phone - as shown in Fig. 2 basically comprises a keyboard 32, a display 31, a control section 30, a radiofrequency section 35, a power supply 33 and an audio conversion section 34 according to the usual components in use. As far as the present invention is concerned, it is important to consider the direct interfacing of the audio section with an equalizer section 37 and that the same audio section 34 supports an LS loudspeaker and two microphones M1 and M2, i.e. high sensitivity acoustic detectors which have minimized transmitted noise.

It actually includes two microphones, ie two transducers for converting the sound signals emitted into electrical output signals in the presence of acoustic noise; moreover, during the conversation the transducer is kept at a substantially constant distance with respect to the source, so that the transducer is able to respond to a second order spatial derivative of the pressure field associated with the input frequencies. As shown schematically in Fig. 2, two differential microphones M1 and M2 are used separated from each other by a certain distance d, in practice an M1 microphone is placed as usual on the lower end of the telephone where the speaker speaks. User, the other M2 directional microphone is mounted at the top at the opposite end, therefore the two microphones, certainly not being directed in the same direction, are able

i- one, M1, to acquire the voice that speaks on the phone plus the surrounding noise while

ii- the other upper one, M2, of directional type, acquires only the noise coming from the surrounding environment.

In the equalization procedure, the signal to be evaluated coming from the loudspeaker must be returned to the input of the sampled and preamplified M1 microphone. It is then filtered by a passband having a passband between 375 and 4000 hertz.

Associated with this pair of differential microphones M1, M2 there are differentiation means that allow to acquire an electrical input signal from each of the two microphones in order to produce, in response to them, an electrical difference signal that is proportional to the difference between the acoustic signals coming out of the corresponding microphones.

The regulation and control principle used by the user interfacing with the telephone network also provides for a real-time detection system that can operate in different modes depending on whether

a- the called user is the person with the least hearing ability,

b- that the calling user is the person with less hearing capacity, and c- that the caller and the called party are both people with impaired hearing.

In the three possible situations there is an interactive management and control of the audiometric adjustment that is operated between caller and called. In a preferred embodiment, the information relating to the fact that caller A or called B are both persons with hearing impairments is acquired by the HLR 10 with this noting that in relation to the call appropriate information must be sent to caller A about type of conversation and the relative modalities to carry out the same. The system therefore provides for the filtering of incoming speech: a- based on the fact that the called person has a reduced hearing capacity on specific frequencies of the voice signal spectrum, b- based on the fact that there is a disturbed communication and if we want to improve its quality, and on the basis of criterion c- that it is better to originally suppress the noise and frequencies that must be sent and which otherwise would be amplified, distorted and further deteriorated.

It is based on the principle that if it is already known that some specific frequencies must not be listened to then it might as well be at the central control level, that is to say at the level of the characteristic parameters of the audiometric curve of the user which is stored in the ™ HLR 10 if filtering and / or amplification is available. Therefore, whenever it is detected that from the calling terminal these frequencies are present in input on the calling telephone, they are filtered to prevent these frequencies from being amplified in amplitude and reported to the called as a perceptible signal.

A further realization provides for the synchronization of the telephones in possession of both users with limited hearing capacity for which an equalization on the two telephones in real time is useful, in relation to noise and specific environmental disturbances. That is to say that the equalization carried out - as described previously - once, may not be the best if you are in a specific more or less noisy environment. In this context, an equalization procedure is envisaged at the very moment in which caller A requests to communicate with a person called B and both the two users, at the mobile telephone network operator, are marked in the relative HLR 10 as persons with reduced hearing ability. Just as a dedicated synchronization would be effective for two people who converse with reduced hearing capacities, frequently by telephone.

In fact, both interlocutors - caller A and called B - of the conversation are offered a procedure for adapting the voice frequencies received and transmitted, in real time and before the actual conversation. The dichotomous equalization procedure, just defined, is established not on the basis of sample frequencies pre-stored at each radio base station or at a distributor shop (specialized center managed by the network operator) of mobile phones, with center functions audiometric, but on the basis of real sample frequencies sent from telephone A to telephone B and vice versa, and acquiring the signal reproduced by the loudspeaker of the first telephone directly detected by the pair of differential microphones M1, M2 same first telephone A bringing it back to the other B which evaluates and compares the spectral component in relation to the fact that a certain pre-memorized standardized frequency component is expected, and instead this arrives deformed, distorted or with a noise component. So in reality it is precisely the loudspeaker of mobile phone A that emits a series of sample sounds, each of which is acquired as a test sound to be sent to the other telephone B which evaluates its quality in real time and considers its quality. ™ any filtering and / or amplification in relation to:

i- to the acoustic sensitivity curve of the listener in possession of the receiving mobile phone;

ii- to the pre-memorized spectral component of reference;

iii- to the predictive and adaptive function which is programmed to consider any improvement on the second mobile phone resulting from the frequency cuts and amplifications that will be brought to the first mobile phone. These same operations will be performed on phone B in a similar way. By inverting the receiver and transmitter, a satisfactory level of speech understanding is achieved without overlapping disturbances.

For example, in order to establish the handshake phase, preliminary to the actual conversation, the first mobile phone reproduces the frequencies 250, 280â € ¦300, .. 800 Hertz etc. according to a pre-established standard sequence, and at the same time the receiving mobile phone. It is synchronized to receive said signals in the predetermined sequence. If a particular frequency is affected on the receiving telephone by deformations or noises, there may be activity on it, through the equalization system a selective filtering. Similarly, if a particular component loses power in the course of voice signal transmission, a selective amplification level is applied to it. It should be borne in mind that the detection of the sample signal takes place in the real listening and reproduction conditions. This is because the signal emitted by the loudspeaker of the transmitting mobile phone A is acquired selectively by the same differential microphone of the mobile phone A which transmits it.

The described equalization procedure is typically adaptive as it is initially considered an intervention on particular frequencies predetermined by the equalization algorithm. In case of failure in a first pass of regulation of the requests and of the frequency components, if a given frequency is still transmitted with significant distortion or noises, then one goes to examine the more or less restricted area of the frequency itself in order to carry out a fine adjustment and avoid that relevant frequency components are cut from the voice signal that is transmitted, consequently limiting substantially the information content of the voice signal.

After a series of localized tests on a particular ∠† f, the components to be lowered / eliminated are empirically selected which would bring more noise while they are enabled for transmission within said interval ∠† f, the cleaner frequencies.

A practical situation exemplifying the flexibility of the present invention could be that of two people with reduced hearing capacities who are talking on the telephone, on the street, near a noisy compressor in continuous action. In this context, the equalization system according to the present invention eliminates the frequencies closest to the noise component produced by the compressor thereby obtaining a filtering on the frequency band typical of this disturbance.

But suppose that over time the noise conditions are variable (for example the compressor is linked to a pneumatic hammer which is only activated at times, producing additional disturbing signals with respect to the starting one), then it is evident that the equalization of the frequencies involved have changed.

The technique according to the present invention provides for a real-time scan to assess whether in fact the choice of equalization initially implemented is suitable for the type of noisy environment in which the speaker is located.

It is well known how the voice is digitized, sampled and transmitted on the mobile phone with sampling in fact very thin. It would even be possible that a value out of two thousand (one millisecond for every two seconds) is sampled to maintain the intelligibility of the speech and lose only at the level of the timbre of the voice. Therefore, it is sufficient to insert in this sampling a detection interval that allows to monitor every ∠† t of time (for example every second) a particular ∠† f and to carry out a check with the receiver according to the procedures already described, on the quality of the voice signal arrived at its destination and from there perform a correction on the equalization level.

Otherwise, the time intervals in which there is no talk are more effectively used. The criterion is always that for which the relative presence of noises perceived by the microphone is detected on a mobile phone and is sent to the telephone with which it communicates for the relative equalization (also in this case by implementing a filtering of background noise).

In fact, therefore, in this solution, sampling is carried out at each word termination. For example, the vocabulary that the mobile phone usually has for writing SMS is used. It is well known that usually in the T9 mode there is a feature on a word active, from time to time, which facilitates the writing of the SMS. If it is assumed that the speech-to-text recognition and conversion function has been set up, a procedure can be activated according to which when it is assumed that the word ends, the detection of the presence of background noise defined as any input signal not attributable to the speech of the user of the mobile phone.

On the other hand, for the purposes of sampling the noise conditions in relation to the speech signal, it was however necessary to carry out an accurate evaluation of the duration of the speech signal in order to make an appropriate distinction between the spoken and non-spoken segments. This evaluation had as its ultimate goal the automatic detection and optimization of the output signal. Moreover, a series of solutions are known which for the resolution of the problem â € œspoken / non-spoken detectionâ € are based precisely on an estimate of the energy of the signal considered. The starting point is always that the energy (and therefore the power) of a spoken segment is greater than the non-spoken one and the fundamental step was the calculation of the absolute value of the samples.

Hence a high sampling reliability deriving from carrying out the sampled detection of the presence of noise at the termination of each vocal segment on the basis of two different synchronization tools such as:

a) the operational phases relating to the automatic detection of the termination of the vocal signal based on the principle that the energy and therefore the power of the spoken segment is greater than the non-spoken one;

b) the presence of a vocabulary already active on the mobile phone that works with an operational functionality upstream of the conversion from speech to text.

In other words, a temporal control is introduced both on the permanence of the vocal signal and on the amplitude of the signal itself. With this mode the comparison between the acquired signal and the background noise is carried out, in the sense that the criterion used is that if the energy of the signal is higher than that detected for the background noise then the input signal can be interpreted. as a vocal segment. This prevents unusually consistent disturbances from being recognized as vocal segments precisely on the basis of the permanence of said signal. The result is that the solution according to the present invention operates a continuous environmental detection carried out on numerous samples not affected by voice but exclusively by environmental disturbances, and from which an extremely reliable average of the spectral components of the present disturbances can be obtained.

The interactive support for listening to the voice signal for the user with hearing impairments is based on a radiotelephone-type operating mode with functions such as â € œpress to speak, release to see the relative textâ € in which the call is activated by pressing a dedicated key on the keypad 32e, which operates as a call switch. By pressing the key, the user indicates that he wants to speak and the terminal prepares to acquire the voice input on the microphone. By releasing the key, on the other hand, it will be possible to communicate to the interlocutor that he is activating a forced pause - synchronization - in the conversation in order to see the relative conversion into a text format on the mobile phone display 31. A solution for the fast conversion to text is constituted by the use of a voice recognizer, now highly reliable and of low cost. For example, speech recognizers are known, independent of the speaker, who use a dictionary of pronunciations in which from a speech based on the word that has been received, from each of them a sequence of text or a sequence of phonemes is obtained. . On the basis of a sequence of phonemes thus interpreted, a text sequence corresponding to the speech is generated which must be transmitted as a text message.

For this solution it is also fundamental to use the SMS server which is set up to convert, in the final analysis, the original vocal signals into SMS messages.

It is known how messages are stored and transmitted by a message center called SMS-Center 15 (SMSC). The message center is the electronic equivalent of the ordinary mail service, as it stores and forwards messages as soon as they can be routed.

It is also known that in order to manage the mobility of users, the MSC component must continuously exchange information with a database, called Visitor Location Register (VLR) 14, which temporarily stores information relating to MS (Mobile Station) A and B which are located in that area (IMEI user identity, MSISDN telephone number, authentication parameters, etc.). The MS in question are simply "visiting" the area served by the VLR 14. In fact, they can move at any time within the area served by another VLR.

In general, it is known that when sending an SMS message the fundamental operations carried out are:

â € ¢ Routing Information Request: First of all, the SMSC 15 needs to retrieve the information necessary for routing from the subscriber database, i.e. from the HLR, in order to determine the cell to which the mobile phone is connected . This process is done before the message is sent

â € ¢ Point to Point Short Message Delivery: This is the mechanism that sends the SMS message data from the Service Center to the MSC, which is currently connected to the cell to which the mobile phone is connected

â € ¢ Short message waiting indication: This operation is activated when the SMSC 15 fails to send the SMS message due to any technical problem. In this way the Service Center requests the HLR to be added to the SMSC list to be informed when the mobile phone becomes accessible again.

â € ¢ Service Center Alert: The HLR 10 informs the SMSC 15 that the mobile phone to which the transmission of an SMS message was previously attempted is now accessible again and can be reached by the MSC.

It is evident that this sequence of procedures is extremely simplified in the application that is being illustrated. In fact, first of all, no â € œRouting Information Requestâ € is necessary as the SMS that reports the conversion from speech to text will be delivered directly to the calling mobile phone, of which the cell in which it is currently operating is obviously known. . Even the â € œShort message waiting indicationâ € has no reason to be implemented as the SMS reporting the conversion from speech to text will be immediately delivered to the caller who is there waiting for the voice communication to be switched. to SMS text. Lastly, the â € œService center alertâ € will also be not activated as there is certainly no need to assess whether the mobile phone is accessible again as it itself has activated a call and is therefore accessible for the entire time of the signal. .

A procedure is thus established in itself which defines - in real time - from â € œwhereâ € (from which cell) and â € œfrom whomâ € the call is taking place, so there is no need to have to interrogate HLR 10 again and / or VLR 14 relating to the identification information of the interlocutors in order to be able to send them the SMS being compiled deriving from the speech component. Furthermore, the text / speech synchronization can be managed â € œmanuallyâ € by the interlocutor who, with the appropriate key, can suspend, forcing a pause.

Another solution in which a combination of the vocal informationâ € "sound with the corresponding in text format is implemented is the one that provides, for GPRS terminals, the creation of chat-like work sessions on TCP / IP protocol , for which contextually to the voice signal transmitted packaged for example in VOIP a conversion of speech into text is defined by means of the already mentioned speech synthesis contextually the visualization on the screen of the PC being used.

In this regard, it is possible to use all the functions already implemented in the relative session layer concerning the logical resources responsible for organizing the conversation between the presentation entities. This in order to structure and synchronize the data exchange so that it can be suspended, resumed and terminated in an orderly fashion. The role of the session layer is, moreover, fundamental since it has the task of masking as much as possible any interruptions in the transport service and in maintaining a logical continuity in the evolution of the session connection.

The voice / text service, managed directly by the user, ensures that the user on the phone does not notice during the communication of any shortcomings or temporary blocks in the connection, the event being perceived at most as a simple delay in the response of the remote system.

Also in the field of TCP / IP GPRS reference is made to any voice recognition technique for converting speech into text.

On the basis of the solutions described, a series of realizations derived from audio equipment such as: intercoms and / or video intercoms, intercoms with one or more channels, headsets, intercommunication headsets (for hands-free activities), guides and voice machines for instructions or for automatic responders.

On the other hand, there are innumerable conditions in which a perfectly gifted person would need a double feedback on the communication in progress, both visual and auditory, and in any case of the highest audio quality. For example, in many military activities, in situations where an operator receives ambient noise through headphones. This is the case of crane operators or drivers of special transport vehicles, or of the conditions in which you want to have more information about those who report being an unidentified messenger, or janitor or more generally, to those who request access to a limited environment, public or private.

The component sections and the functionalities carried out for these last applications are basically the same as described up to now. The effectiveness of using a double microphone that differentiates the incoming audio signals in order to separate the actual vocal component from the incoming random disturbances is evident. Similarly, the use of a voice recognizer is fundamental that translates the words spoken in real time into a relative written text with the possibility of viewing - between the two interlocutors - synchronized through a device such as â € œpress to view, release to speakâ € .

It is also fundamental to regulate and filter the frequencies in real time, dynamically amplified, through a dedicated menu.

At the same time, the use of algorithms is required, which allow to distinguish and effectively predict the speech phases from the non-spoken ones in order to carry out real-time sampling of the environmental noise conditions, even though these may be random and subject to environmental disturbances of various kinds.

The resolution of the problem â € œspoken / unspoken detectionâ € is still based on an estimate of the energy of the signal considered. The starting point is always that the energy (and therefore the power) of a spoken segment is greater than the non-spoken one. The sampled detection of the presence of noise at the termination of each vocal segment is carried out on the basis of two different synchronization tools:

a- the operational phases relating to the automatic detection of the termination of the vocal signal based on the principle that the energy and therefore the power of the spoken segment is greater than the non-spoken one;

b- the presence of a vocabulary already active on the interconnection device which interacts with an operational functionality upstream of conversion from speech to text.

The same use of the GSM / GPRS telecommunications networks, already described, can be easily integrated in the various constructions mentioned last; just think of an intercom or intercom that instead of working with a single direct or switched cable, for interconnection, interposes a whole connection in TCP / IP with the establishment of two concurrent sessions such as VoIP and Chat, between the two interlocutors; the complexity of the solution while still allowing to have the advantages of textual feedback corresponding to speech. Advantages and industriality of the invention

In summary, the prerogatives of the conversation equalization techniques according to the present invention are of fundamental help for the person with limited acoustic abilities, since they present:

- maximum adaptability to the user's hearing needs;

- the highest quality of sound reproduction and complete reprogramming with a different landline / mobile phone;

- the range of application for all levels of hearing loss: mild, medium, severe and profound level.

The cellular telephone according to the present invention is a real expert system capable of detecting the presence of the voice and consequently adapting the amplification modes of the apparatus. The equalization technique, of the audio signal in fixed and mobile telephony, of the invention according to the present invention can be implemented not only on the basis of classical audiometric reports (audiometry, speech, impedance, etc.) but also on the basis of the processing capabilities of the sounds from the patient, as well as it can be detected at the moment of the conversation in real time. This improves speech understanding in the presence of noise and reduces acoustic overstimulation in the absence of speech. This last possibility is particularly important to reduce the acoustic fatigue of the elderly who frequent very noisy environments.

It will be understood that the above description and the accompanying drawings are intended only to illustrate the present invention. It is obvious to those who are well versed in the field that the invention can be varied and modified in other ways as well without departing from the fundamental purpose of the invention as defined by the attached claims.

Claims (1)

CLAIMS 1. Technique of equalization of the voice signal that is transmitted through a telephone channel, characterized by carrying out at the receiving telephone level (B), a processing procedure that evaluates the composition of the frequency spectrum of the signal received by means of the transmission channel in relation to the spectrum of the original signal transmitted by the calling telephone (A) and by performing the processing and equalization of the received signal through real-time filtering of the frequencies making up the spectrum to be followed by the selective attenuation of some frequencies and the selective amplification of others, in relation to the current environmental noise conditions and the hearing qualities of the receiver 2. Technique of equalization of the voice signal that is transmitted through the telephone channel according to claim 1, characterized by the fact of providing a procedure for analyzing and processing the signal in real time, each time a communication is established, the equalization being able to adapt to the specific environmental conditions of the moment. 3. Technique of equalization of the voice signal that is transmitted through a telephone channel according to claim 1 characterized by the fact of providing that at the level of each telephone the signal processing procedure that intervenes on the composition of the spectrum is based on an established parametric configuration based on the auditory characteristics of the user, performed one-off, at the time of the initial configuration of the telephone 4. Technique for equalizing the voice signal transmitted through the telephone channel according to each of the preceding claims, characterized by the fact that both calling interlocutors cooperate ( A, B) of the conversation, in a procedure for adapting the voice frequencies received and transmitted, in real time and before the actual conversation. 5. Technique of equalization of the voice signal transmitted through telephone channel according to each of the preceding claims, characterized by the fact of providing for the synchronization of the telephones in the possession of users both with limited hearing capacities by arranging the equalization on both telephones in real time, in relation to the noise and specific environmental disturbances of the particular communication in progress, filtering and amplifying certain frequencies of the speech, already in input, to the calling telephone (A) in relation to the fact that the communication takes place in conditions of noise and that the called person has a reduced hearing capacity on specific frequencies of the speech signal spectrum. 6. Technique for equalizing the voice signal transmitted through a telephone channel according to each of the preceding claims, characterized by providing an aid to the hearing-impaired user both by arranging a real-time equalization that intervenes on the basis of the environmental noise conditions on particular frequencies, both by providing a visual aid in textual form that translates the voice signal into sequences of words in text format that can be interactively displayed on the mobile phone display. 7. Equalization technique of the voice signal transmitted through the telephone channel according to each of the preceding claims, characterized by the fact of carrying out a sample detection in real time, in the same telephone call by carrying out the random sample detection of the presence of noise at the termination of each voice segment based on two different synchronization tools: i- the presence of a vocabulary already active on the telephone which interacts with an operational functionality upstream of the conversion from speech to text; ii- the automatic detection of the termination, word by word, of the vocal signal based on the principle that the energy and therefore the power of the spoken segment is greater than the non-spoken one; hence introducing a temporal control both on the permanence of the voice signal and on the amplitude of the signal itself, based on the principle that only if the energy of the signal is higher than that detected for the background noise then the input signal It can be interpreted as a vocal segment. 8. Technique for equalizing the voice signal transmitted through a telephone channel according to each of the preceding claims, characterized by the fact that the aid based on the interactive display of a text corresponding to the voice signal for the user with hearing impairments is managed with the functions type â € œpress to speak, release to see the relative textâ € in which the call is activated by pressing a dedicated key on the keypad (32), which operates as a call switch, with the pressure or release of the key being made synchronization in the conversation is possible, which allows you to see the relative conversion of speech into text format on the telephone display 31. 9. Technique for equalizing the voice signal transmitted through the telephone channel according to each of the preceding claims, characterized by the fact of implementing the combination of the voice-sound information with the corresponding one in text format, through the use of GPRS terminals with which an instant messaging activity such as chat or instant messagining on TCP / IP protocol is also implemented together with the speech communication function, so that, at the same time as the voice signal transmitted packaged in VOIP, a conversion of speech into text is carried out at the same time by means of voice recognition and its transmission on a dedicated concurrent session. 10. Equalization apparatus for the voice signal transmitted through a telephone channel according to each of the preceding claims, characterized in that it uses a telephone device equipped with a high-sensitivity acoustic detector comprising two microphones (M1, M2) with relative transducers for converting the sound signals emitted in electrical output signals separated from each other by a certain distance (d), such that one (M1) acquires the voice speaking on the telephone plus the surrounding noise, while the other (M2) of directional type acquires only the noise coming from the surrounding environment, associating to the pair of differential microphones (M1, M2) differentiation means that allow to acquire an electrical input signal from each of the two microphones in order to produce, in response to the same, a difference signal that is proportional to the difference between the acoustic signals output from the corresponding microphones. 11. Technique of equalization of the voice signal transmitted through telephone channel according to each of the preceding claims, characterized by the fact of implementing an equalization of the signal based on the real sample frequencies sent from the first telephone (A) to the second telephone (B) and vice versa , and acquiring the signal reproduced by the loudspeaker of the first telephone (A) directly detected by the pair of differential microphones of the same first telephone (A) and reporting it to the other (B) which evaluates and compares the spectral component in relation to the the fact that a given pre-memorized standardized frequency component is expected, which may arrive instead distorted or with a noise component. 12. Technique of equalization of the voice signal transmitted through the audio channel according to each of the preceding claims, characterized by the fact of carrying out the combination of the voice-sound information with the corresponding one in text format, in devices and apparatuses of the audio type such as interphones and / o video intercoms, intercoms with one or more channels, headsets, intercommunication headsets, guides and voice machines for automatic responders, even in the conditions in which a perfectly equipped user needs double feedback on the communication in progress, both visual and auditory, and however of the highest audio quality, and in which you want to have more information about the person requesting access to a limited area, public or private, by storing the sequence of scanned texts in the speech-to-text conversion in order to determine the identification of the person and the authenticity of the information provided.