Patent Publication Number: US-11051115-B2

Title: Customizable audio signal spectrum shifting system and method for telephones and other audio-capable devices

Description:
CROSS REFERENCES TO RELATED APPLICATIONS 
     This application claims the benefit of the U.S. provisional patent application 62/921,601 filed on 26 Aug. 2019 and titled Audio Signal Spectrum Shifting System and Method for Telephones and other Audio Capable Device This application is also related to the U.S. provisional patent application 62/921,601 filed 27 Jun. 2019 and titled Customizable Audio System and Method for Telephones and other Audio Capable Device. 
    
    
     COPYRIGHT NOTIFICATION 
     Portions of this patent application contain materials that are subject to copyright protection. The copyright owner has no objection to the facsimile reproduction by anyone of the patent document or the patent disclosure, as it appears in the Patent and Trademark Office patent file or records, but otherwise reserves all copyright rights whatsoever. 
     TECHNICAL FIELD 
     The present invention relates to methods and systems for improving quality of audio communications as audibly perceived by humans. 
     BACKGROUND 
     A significant percentage of the population is experiencing various hearing problems. While some of their needs are addressed by conventional hearing aids, the hearing aids are often insufficient or cumbersome, particularly when used during telephone conversation. Furthermore, people are not always wearing hearing aids and often have to take a phone call while not wearing a hearing aid. This creates a need for a telephone that would provide hearing assistance during a phone call independently of other means. Also, people without hearing problems have different acoustic preferences while listening to audio. This creates a need for a customizable audio system for use in telephones, audio-capable computers, and other computing and communication devices such as radio telephones and walkie-talkies, as well as aviation transceivers. 
     Conventional telephone audio systems usually address audio challenges through amplification. 
     Audio signal perception by a human user is a function of two components: human user&#39;s hearing and amplification by the device&#39;s audio system. However, human user&#39;s hearing loss in some part of the audio spectrum sometimes is so severe that available amplification cannot bring the signal volume to an acceptable level. This problem usually pertains to high end of the audio spectrum. Unfortunately, the high end of speech audio spectrum contains a lot of information that is essential for speech recognition. This is particularly evident when listening to a speaker with a high-pitched voice. In such cases, while the overall volume of the audio signal may be very high, the speech recognition may be severely damaged. In other words, amplification of low end of audio spectrum may be insufficient if the high end of the spectrum cannot be heard. This is particularly important for telephone speech communication, where the speech recognition may be an important part of the audio system&#39;s usefulness. 
    
    
     
       BRIEF DESCRIPTION OF FIGURES 
         FIG. 1  is a general functional diagram of the audio system according to a preferred embodiment of the present invention. 
     
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
     Detailed embodiments of the present invention are disclosed herein. However, it is to be understood that the disclosed embodiments are merely exemplary of the invention that may be embodied in various and alternative forms. Specific structural and functional details disclosed herein are not to be interpreted as limiting, but merely as a representative basis for the claims and/or as a representative basis for teaching one skilled in the art to variously employ the present invention. 
     Proposed System 
     The proposed system  10  is based on audio signal spectrum frequency shift. Hearing loss on some frequencies could be so significant that a combination of available amplification and output transducer parameters is still not sufficient to provide acceptable quality of audio signal for the customer. For example, this can happen with common age-related hearing loss on high audio frequencies. This effect can be particularly undesirable because often higher frequencies contain significant audio information important for speech recognition. In such a case, even with acceptable amplification in the lower part of the audio spectrum, speech recognition can be impaired. To address this challenge, the proposed system includes frequency spectral “shift,” or frequency re-scaling. This shift can be performed by shifting the entire input audio signal to a different part of the audio spectrum [See sources 1-3, 5]. This can be done similarly to using a musical instrument and playing the same melody, for example, one octave lower. This shift can preserve much of the information in the input audio signal and thus present it in an audio spectrum that is better heard by a customer. 
     Also, this could make the “shifted” signal subject to easier amplification requirements, thus producing an audio signal for better speech recognition by customers with severe frequency loss in the high end of the audio spectrum. When customer&#39;s initial audiogram [4] shows severe high frequency loss that cannot be remedied by the combination of available amplification and output transducer parameters, the system executes the described frequency spectrum shift of the incoming signals that contain significant high frequency component. 
     In other words, it repeats volume of incoming signal&#39;s every detected audio frequency on a different corresponding lower frequency. The corresponding frequency is determined by a chosen coefficient, for example 1/2. This is done in a manner similar to a technique known in music industry as “pitch shift” [3]. The mentioned ratio 1/2 would become an equivalent of playing music melody one octave lower. 
     The proposed solution addressing the problem described above is represented by system  10  and method describe below. The system can be built in different implementations. It can be autonomous, i.e. built into a conventional telephone or other audio-5 capable communications devices. Alternatively, some of the system&#39;s functions can be implemented in a remote facility such as a carrier&#39;s or a provider&#39;s server or a phone manufacturer&#39;s service center which can be accessed remotely. Such implementation can provide some of the services described below in a remote mode. 
     The system&#39;s general functional diagram is depicted in Drawing  FIG. 1 . 
     Components of the System  10  (i.e. Explanation of the Reference Numerals of  FIG. 1  and of the Method of Operation) 
     The system includes a Spectrum Shift capable Receiver  1  operable to receive an audio signal from communications line or other source through the telephone, computer, or other audio-capable device, including input transducer such as a microphone. Receiver  1  is capable of audio signal frequency spectral “shift,” or spectral re-scaling, upon receiving a command from the Spectrum Shift Controller  5  or from manual command from a human user, as described below. 
     The system  10  includes an amplifier  2  electrically connected to the spectrum shift capable receiver  1 , the amplifier  2  receiving an electronic form of audio signals from receiver  1 . The amplifier  2  performs signal amplification in accordance with pre-programmed instructions and provides it to the Audio Output Transducer  3  such as a telephone speaker or headphones. 
     The system  10  includes an audio output transducer  3  (such as a speaker, headphones, or a similar device), usually a part of a telephone or other audio-capable device), the audio output transducer  3  operable to receive electronic signals from Amplifier  2  directly, or through wired or wireless connection such as Bluetooth. The audio output transducer  3  is operable to convert received electromagnetic signals into audio signals to be heard by a human Customer C. 
     The system  10  includes a hearing tester/calibrator  4  in electrical communication with the amplifier  2  and operative to administer automated speech recognition tests through audio output transducer  3  and, ultimately, to the customer (user) C. Based on the tests results, it determines frequency threshold for acceptable speech recognition for user C through Transducer  3 . The goal of the test is to determine a volume level for the highest frequency needed for acceptable speech recognition. The hearing tester/calibrator  4  provides the determined value of the threshold to a spectrum shift controller  5  for decision and implementation of the audio signal spectrum shift. This type of test has two advantages over a conventional audio test: it is automated (i.e. does not require a visit to a specialist), and it also includes a specific output transducer, thus addressing variations in the transducers&#39; parameters. The later may improve accuracy since it tests human hearing with a specific transducer to be used in the device, vs. a generic transducer used in a conventional audio test. Functions of the hearing tester/calibrator  4  can be implemented in the telephone autonomously, or remotely from a centralized location such as manufacturer&#39;s or service provider&#39;s server, or it can be partially implemented  5  at both locations. 
     The spectrum shift controller  5  determines if an amplified audio signal from amplifier  2  is sufficient for speech recognition in of the audio spectrum, particularly in the high-frequency end of the spectrum. If the signal is not sufficient, it gives a command to receiver  1  to “shift” the entire incoming signal spectrum toward a lower frequency. 
     A telephone/audio-capable computer  6  is a host device for the system  10 , such as a telephone, an audio-capable computer, or other computing and communications device such as a radio telephone, a walkie-talkie as well as an aviation transceiver. It interacts with the proposed system  10  by forwarding the incoming signal, and also providing means of control such as a dedicated button, or use of existing buttons for turning spectrum shift on. 
     A human user/customer is referred to with reference character “C” and is expected to turn the parameter of the spectrum shift on and off by use of a dedicated or existing button of the telephone or other audio-capable device. It is capable of overriding a spectrum shift command from the spectrum shift controller  4 . It is understood that a human person is not actually a component of the invention but intended to participate in the method of using the system  10 . 
     Modes of Operation 
     The system functions in two modes: 
     1. Initial Calibration/Subsequent Adjustment 
     2. Normal Operation, i.e. providing customized fine-tuned optimized audio signal to customer 
     1. Initial Calibration/Subsequent Adjustment mode—the hearing tester/calibrator  4  administers automated speech recognition in initial configuration of the telephone or other audio-capable device. The initial test is similar to a common speech recognition test provided by audiologists. The test determines the minimal required signal volume at the highest frequency necessary for the incoming audio signal that is necessary for acceptable speech recognition by Customer C receiving the signal through transducer  3 . This test is performed at the initial calibration and can also be used later for fine-tuning, allowing for flexibility of addressing possible hearing changes of a customer in time. Also, this telephone/audio-capable computer  6  can allow adjustment and programming of the amplifier  2  parameters in a case of changing the transducer  3 . It should be noted that, for convenience purposes, a customer may want to calibrate the system to more than one transducer. This would require having more than one amplification schedule in amplifier  2 . The system  10  can be programmed to choose a particular amplification schedule automatically in accordance with a particular transducer used at the time. Also, this system  10  allows the phone to be customized for subsequent users. Further yet, this system  10  can be expanded to include customization for several customers (such as in a family use) with every customer turning on a specific customization upon taking up the phone. 
     2. Normal Operation—The hearing tester/calibrator  4  is turned off and the system  10  automatically fine-tunes received audio signals in accordance with a customer&#39;s specific individual needs, with a frequency spectrum “shift,” when it is determined necessary for acceptable speech recognition. The system  10  can have a manual override. For example, if the system  10  does not perform spectrum “shift” automatically, the shift can be ordered manually by customer&#39;s command, for example by pushing a dedicated or programmed standard button of the phone. Also, an automatic shift can be cancelled manually if the customer does not recognize its usefulness. 
     It should be noted that the system  10  can be used without a communications line. In this case it can act as a hearing aid, receiving and amplifying audio signals through its input  20  transducers such as microphones (not shown in  FIG. 1 ). 
     The proposed system  10  can be built in an autonomous implementation, when it is fully contained in a telephone or another audio-capable device. It can also be built in a centralized implementation, when some or all functions of the system are placed in a remote location such as a communications carrier&#39;s or a phone manufacturer&#39;s service center. 
     Regarding the invention being thus described, it will be obvious that the same may be varied in many ways. Such variations are not to be regarded as a departure from the spirit and scope of the invention, and all such modifications as would be obvious to one skilled in the art are intended to be included within the scope of the claims. It is to be understood that while certain now preferred forms of this invention have been illustrated and described, it is not limited thereto except insofar as such limitations are included in the following claims. 
     REFERENCES 
     
         
         [1] Mark Dolson. The phase vocoder: A tutorial. Computer Music Journal, 10(4):14-27, 1986. 
         [2] J. L. Flanagan and R. M. Golden. Phase vocoder. 5 Bell System Technical Journal, pages 1493-1509, November 1966. 
         [3] Jean Laroche and Mark Dolson. New phase vocoder technique for pitch-shifting, harmonizing and other exotic e ects. In IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, N.Y., 17-20 Oct. 1999. 
         [4] Frederick N. Martin and John Greer Clark. Introduction to Audiology. Pearson, 12th edition, 2014. 
         [5] M. R. Portno . Implementation of the digital phase vocoder using the fast Fourier transform. IEEE Trans. Acous., Speech, Sig. Proc., 24(3):243-248, June 1976.