JPH06175817A - Test and compensation for hearing acuity of multimedia - Google Patents

Abstract

PURPOSE: To compensate the voice output of a multimedia system so as to effectively utilize all the effect of a multimedia program while allowing a user to easily distinguish a tone. CONSTITUTION: A system and method is used for detecting and compensating the hearing disorder of a testee. First, a hearing test generated by the multimedia system is given to the testee. Next, a voice frequency which is outputted from a multimedia computer system and in which the testee hearing acuity is recognized to be disabled through the use of the result of the hearing test is increased/reduced so as to properly compensate it. In the environment of a class room, the result of the hearing test can be used by the testee or a teacher.

Description

Detailed Description of the Invention

[0001]

FIELD OF THE INVENTION This invention relates generally to multimedia systems for identifying and correcting hearing impairments in users. More specifically, the user is subjected to a hearing test to identify tones that the user is unable to hear properly.
The output of the multimedia system is then compensated based on the user's individual hearing profile.

[0002]

2. Description of the Related Art Many people have impaired hearing, that is, the ability to distinguish sounds. Generally, these disorders have physical or medical causes that result in the inability to distinguish a range of sounds. Usually, unless the hearing is completely lost, increasing the sound level (volume) can compensate for these obstacles. Therefore, many people wear hearing aids that they place in their ears to amplify the sound. Hearing loss is gradually lost, so hearing loss is often unnoticed until the lesion has progressed considerably. Younger children often do not notice their hearing impairment, which can result in poor school performance. Such children may not be able to properly hear the teacher's speech due to hearing impairment for certain ranges of audible frequencies.

Hearing tests are usually performed on those who wish to work in workplaces with high noise levels. A hearing test is also conducted when hearing loss occurs due to an old age or accident. These hearing tests are used to determine if a hearing aid is needed and how the hearing aid should be set up.

For example, US Pat. Nos. 3,809,081 and 4,321,427 describe devices and methods for conducting hearing tests. US Patent No. 4953112
Issue is a software model of the hearing aid, again testing the user under various conditions without using the hearing aid itself. U.S. Pat. No. 4,768,165 is a system for controlling audiometer files to enable correction of hearing aids. Similarly, U.S. Pat. No. 4,489,610.
No. is also a computerized audiometer, which analyzes the test data and creates a program that describes the amount of correction required by the subject. Furthermore, this program is used to set the hearing aid to amplify the received voice within a certain frequency band. That is, a computer program is used to adjust the response of the hearing aid to the received voice.

[0005]

Presently, multimedia computer systems are used in many environments such as classrooms and vocational training. In particular, in a classroom environment, it would be highly advantageous to have a multimedia system that could be tested to determine if a student has a hearing impairment, that is, if it could interfere with learning.
This information could be used to inform teachers that a student needs to be seated in the front row to hear better. You can also create a software user profile by testing, which can be used to compensate the output of multimedia presentations according to the profile of a particular student to get the full multimedia effect. Will be able to.

[0006]

SUMMARY OF THE INVENTION In contrast to the prior art, the present invention provides a multimedia system that allows a user to easily distinguish between tones and take advantage of all the effects of a multimedia program. To compensate the audio output of.

A system and method for detecting and correcting a hearing problem in a subject is used. First, the subject is subjected to a hearing test generated by the multimedia system. The results of the hearing test are then used to appropriately increase the audible frequency at which the subject's hearing is found to be impaired. Further, in a classroom environment, the results of the hearing test can be used by the test subject or the teacher.

[0008] Generally speaking, a subject starts a hearing test program that outputs tones to the left or right channel (corresponding to the left or right ear) of a set of headphones at a very low volume. The tones that are output make up the entire audio test range, but are selected from a randomly ordered list so that the subject cannot predict the next tone. The list is
It may be used twice, once for the left and right ears, or may be used for both ears, with the tones randomly combined to form a single list. The subject responds to the tone using an input device connected to the multimedia system, such as a keyboard, mouse button, or touch screen. If the system does not receive a response within a predetermined time (eg, 1 second) after the tone is output, the volume is increased slightly. Continue this until the subject responds or the volume reaches the maximum level. The test results for the particular tone are then saved and the volume reset to the lowest level. The next tone is then output and the process continues until the test is complete.

Upon completion of the hearing test, a user profile is created that provides an offset factor for each frequency range tested. Typically, the offset factor has the additive property of increasing the volume to compensate for hearing impairment. Because a certain tone is uncomfortable for the subject,
Tones must be reduced and the offset can be negative. In such cases, the offset factor of the user profile is used to modify the audio output of the multimedia computer system in real time.
The modification of the multimedia output allows the user to hear, understand and experience the program clearly.

Therefore, based on the above summary of the present invention, it will be apparent to those skilled in the art from the objects, features, and advantages of the present invention when examined in combination with the following description and the accompanying drawings.

[0011]

1 illustrates a typical data processing system that can be used with the present invention. Central processing unit (CPU)
10 is 1 of Intel (Intel) X86 processor
Can be one. CPU 10 is interconnected to various other components by system bus 12. The read only memory (ROM) 16 is the system bus 12
It includes a basic input / output system (BIOS) connected to the CPU 10 via the and controlling basic computer functions. Random access memory (RAM) 14, I / O adapter 18, and communication adapter 34 are also interconnected to system bus 12. The I / O adapter 18 may be a small computer system interface (SCSI) adapter that communicates with the disk storage device 20. Communication adapter 34 interconnects system bus 12 with an external network that enables the data processing system to communicate with such other data processing systems. Input / output devices are also connected to the system bus 12 via the user interface adapter 22 and the display adapter 36. The keyboard 23, track ball 32, mouse 26, and speaker 28 are all interconnected to the system bus 12 via the user interface adapter 22. The display monitor 38 is connected to the system bus 12 by the display adapter 36. Further, an audio card 23 such as an audio conversion adapter (MACPA) commercially available from IBM is interconnected with the system bus 12 so that other system input / output such as voice input through the microphone 21 or voice output to the headphones 27 can be performed. Will be provided. It should be noted that audio card 23 provides much greater voice capability than that provided by user interface adapter 22. For example, audio card 23 provides two-channel (stereo) input / output rather than monophonic sound, such as a beep, generated by speaker 28 by user interface adapter 22. In addition, audio card 23 includes a digital signal processor (DSP) capable of processing digitized audio signals in real-time applications. The audio card 23 will be described later in detail with reference to FIG.

FIG. 2 shows a graph of the hearing range where the loudness of an average person is equal. Frequency is plotted along the X-axis, with volume or frequency represented on the Y-axis. It should be noted in the graph that the volume for a person with normal hearing varies as the frequency changes from about 20 HZ up to about 20 KHz. Needless to say, each individual's "standard" volume level for tones of different frequencies is slightly different, but Figure 2 shows various hearing tests and studies conducted to create an "average" hearing range. Represents a statistical summary of.

In contrast, FIG. 3 is a graph of a person with hearing impairment for a constant tone at a particular frequency. For example, the subject in the graph of FIG.
And have hearing loss at 1 KHz.
More specifically, when the threshold of hearing is 100% of the value plotted on the curve (200H in this example).
Approximately 35db for both z and 1KHz) is considered normal, point A represents hearing loss at 200Hz and point B is 1K.
Note that it represents the impairment in Hz. Hearing tests performed on a person to create the graph of FIG. 3 also determine the percentage of that person's hearing loss or strength at various frequencies. Briefly, the test involves entering a tone volume (in decibels), which is a specific frequency, and incrementing the volume until the user responds that the tone is heard. It is known that the average person hears a signal tone at a certain percentage (usually expressed in decibels) for a volume of 100% (at a given frequency). For example, if it is known that the subject does not respond until the volume level reaches 50 db, and the average person responds at 40 db, then 10 db is the amount of hearing loss for a tone of a given frequency. Therefore, the hearing compensation coefficient can be obtained based on the volume disturbance. The hearing compensation factor is equal to the decibel ratio of the volume of the tone actually heard by the subject divided by the volume level heard by a normal user. That is, hearing coefficient = hearing volume / average volume. The decibel is a logarithmic function, and the actual ratio is (10exp (heared volume / 20) / (10
Exp (normal volume / 20)). In this example, for a person who had “normal” hearing in the test, the hearing compensation factor is equal to 3.16 (this value is used by the average user to determine the volume level at which the user actually responded). You need to multiply the volume level to respond).
These hearing compensation factors are values greater than 1 and when multiplied by the volume level at a given input frequency, the volume level is increased by the amount necessary to bring the volume level to 100% normal hearing at that particular input frequency. To increase. This method of determining hearing impairment will be described in detail with reference to the flow charts of FIGS. The present invention determines the percentage of hearing impairment of a subject at various frequencies, and then performs real-time modification of play files in the multimedia system to allow the user to fully utilize the multimedia system and limit hearing impairment. Systems and methods for avoiding disability.

In the multimedia system of the present invention,
A set of high quality headphones is needed to provide the user with an accurate test. Therefore, the detection circuit of FIG. 4 can be used as an initial step in a hearing impairment test to interconnect a set of suitable headphones 27 to a multimedia system.

In particular, there is a problem when connecting headphones to a multimedia system that normally uses speaker output. In multimedia systems, there may be no indication that headphones are being used instead of speakers. For this reason, a signal with a large amplitude may be output to the headphones, and the sound pressure level may increase.
There is also the potential for safety hazards. Different types of headphones have different responses. That is, the loudness can vary by type and brand at various frequencies. For this reason, hearing tests need to be calibrated to the particular type and brand of headphones. It is possible to create a calibration file for widely used brands of headphones and include it as part of the present invention. However, it is not possible to create a calibration file for all headphones on the market, so the present invention includes a test function to determine if compatible headphones are connected to the multimedia system. is there. Therefore, the speaker output driver (RHout and L on the multimedia system
Hout) was determined to provide a maximum effective (RMS) voltage that was too high for most headphones to handle. The solution to this problem is the headphone sensor circuit shown in FIG. The headphone sensor circuit detects when a pair of headphones is inserted and automatically lowers the volume by limiting the maximum power output available from the amplifier for each channel.

FIG. 4 shows each channel (left side and right side).
Two identical circuits for are shown. These circuits can be used to detect monophonic headphones or stereo headphones. (Each about 18 kΩ
To 27kΩ, 22kΩ to 33kΩ)
Resistors R7 and R8, (0.1 μF to 0.5 μ
A common source reference voltage circuit 100 is shown with a capacitor C3 (in the F range). This RC circuit has a reference potential of 3
V and is used to set the threshold voltage level for negative inputs on the right channel compare mechanism 102 and the left channel compare mechanism 104. Comparison mechanism 102, 10
4 is a commercially available product such as the LM339 model. The right and left channel headphone connectors are connected to the right channel comparison mechanism 102 and the left channel comparison mechanism 10 via large time constant RC circuits 106 and 108, respectively.
4 positive inputs. Large time constant RC circuit 10
6 and 108 function as filters to prevent the audio signal from malfunctioning the sensing circuit of FIG. Normally, when the headphones are not interconnected to the sensing circuit, the positive comparator input voltage remains above the reference voltage (provided by the common source reference voltage circuit 100), and thus the comparator outputs a negative active signal. A high level can be maintained by outputting Hsense (headphone detection). Conversely, when the headphone set is inserted and connected in the sensing circuit, the voltage at the positive inputs of the comparators 102 and 104 drops below the reference voltage, which also causes the outputs (Hsense) of the comparators 102 and 104 to drop. Follow up to a low level.

More specifically, the headphone detection circuit is
At the positive input to the comparator (between power and ground):
It has a bias point between the power supply and the ground. In the circuit shown in FIG. 4, between the power source and ground, that is, the comparison mechanism 102,
There is 12V across 104, so the bias point is 6V. This bias voltage provides a means of detecting when a set of headphones is not connected to the sensing circuit. This is because the power supply voltage applied to the comparison mechanism is maintained at the midpoint (for example, 6V). However, when the headphones are connected, the low impedance of the headphones (as opposed to the open condition when the headphones are not present) (about 600Ω) divides the input voltage of the comparison mechanism below 0.7V. Therefore,
An input voltage of 0.7V (present when headphones are present) is present (present when headphones are not present) 6
Since it is below V, the Hsense signal is activated and the presence of headphones is detected.

The flow charts of FIGS. 5 and 6 are the process steps used to detect a user's hearing impairment in the present invention. The process begins in step 1 and in step 2 the presence or absence of headphones is tested as described above for the circuit of FIG. In step 3, the test system sets the tone file address pointer to zero. That is, the system is effectively initialized. A tone file is a list of tones that make up an audiometric test and, as mentioned above, contains tones that can be output on the right channel or the left channel and randomly ordered. The list can be used twice (once for each channel) or the tones for both channels can be combined to create a single 2-channel list and the tones on the list can be used only once. In step 4, the system retrieves the first address for the first tone in the file.
It then receives the tones from the tone file. Also,
In step 4, the initial volume of the received tone is set to zero. The tone is then output on one channel at a volume equal to the default volume (zero). This default volume is actually the edge of the noise level (the tone is heard clearly separated from the background noise). For example, if you are using a system with a signal to noise (S / N) ratio of 76db and the noise level is between 10db and 30db:
The volume is 30d to compensate for the upper range of the noise level.
Start with b. S / N ratio of 76d at the first volume of 30db
When b is added, the volume reaches 106 dB, which is the maximum. Discomfort and distress thresholds are usually 120db to 140d
It is a range of b. In step 5, the present invention continues to increment the volume by 2db until the user responds. For example, the initial volume level is 30db, and the subsequent volume is 32d.
b, 34db, 36db. ． ． , 106 db (maximum). That is, volume = volume (initial) +2 db.
Thus, in this example, including the initial volume level of 30 db and the final volume level of 106 db, there are a total of 34 different volume levels available for testing the user (subject). Therefore, from FIG. 3, it can be seen that the volume shortage of the user at the point B in the graph is 50 db-36 db = 14 db. Similarly, the volume loss of the subject at point A is
66db-36db = 30db.

In addition, the present invention uses decibels as a measure in determining frequency, or tone. In the preferred embodiment of the invention, tones are generated every third octave and are calculated by multiplying the octave start frequency by a cube root of two. Therefore, the tone
The tones stored in the list range from about 31.25 Hz to 2
The range is up to 0.159 KHz. Using this tone selection method, 28 tones are produced at incremental frequencies, as described above. Using 28 tones (one for the left and one for the right ear, for a total of 56 tones), and assuming each test takes 30 seconds (this time depends on user reaction time and various Includes time to increment tone to volume level, test less than 30 minutes (approximately 28
Minutes). Of course, if the subject has impairments in multiple frequency ranges, the test will take longer. Similarly, the test can be completed in less time if the subject's hearing is extremely sharp and responds at a level that is inaudible to the average person. The duration of this hearing test is very important, especially when testing young children. I mean,
This is because if the test takes too long, the test subject's attention and concentration will decrease. After outputting the tone at the specified volume, the system waits for about 1 second (or other suitable time) for the subject's reaction from the I / O device in use. Step 6
Then, it is determined whether or not a response is received from the subject within the designated time, and if it is received, the process proceeds to step 7. However, if no response is received from the user, the system proceeds to step 12 and determines if the volume level is at maximum level. If it is determined in step 12 that the volume is not maximum, the process returns to step 5 and the volume is further increased by 2 db.
Increment. These iterative steps continue until a response is received from the subject or the volume reaches a maximum acceptable level, eg 106 db in the above example. The tone frequency and the corresponding volume, which is indicated by user input that a response has been received, is saved in the user profile (step 7). When it is determined in step 12 that the subject does not hear a tone, in step 13, the tone frequency at the initial volume level is saved. If there is no response at 100% volume tone, it indicates that the user has no tone at that particular test frequency, and the present invention compensates the multimedia system to assist the user in this frequency range. Note that this cannot be done. Following steps 7 and 13, the hearing test process determines whether the tone currently being tested is the last tone in the tone file. If it is the last tone, proceed to step 9 and create a user profile consisting of tone frequency and volume level. Note that there is one tone and volume entry in the user profile for each tone (which is output on both the right and left channels) in the tone file. This user hearing profile table contains cumulative data for the entire test, including offset factors for each frequency range tested. For example,
If the user's hearing is impaired for a particular tone within this range, the offset factor is positive and an addition factor is generated for inclusion in the profile for this frequency.
More specifically, using the above method, if the average response for a particular frequency range is a volume level of 36db, and the user responds at a volume level of 66db, then using the above equation, 31. A factor of 6 (ie 10 exp 66/20 to 10 exp 36/2
(Divided by zero) should be multiplied by the volume level that the average user would react to to compensate for the user's hearing impairment at that frequency. After creating the user profile, the process ends at step 11. However, if it is determined in step 8 that the tone under test is not the last tone in the tone file, then in step 10,
Increment the tone address pointer to test the next tone in the file (new tone address = current tone.
Address + 1, address = N + 1), the hearing test process returns to step 4 to retrieve a new tone.

FIG. 7 is a block diagram showing the components that need to be included on the audio card 23 of the present invention to modify a multimedia play file to compensate for a user's hearing impairment. More specifically,
Two identical circuits are shown for the left and right stereo channels on the multimedia audio card. Note that both circuits have the same components and the same functions in the same way. Therefore, only one circuit will be described in detail. Those skilled in the art will appreciate that the two circuits 2
It will be clear that 03 and 203a can be considered identical. Further, the components of circuits 203 and 203a will be designated with the same reference numbers. However, "a" is added to the circuits and components representing the right channel.

When an analog signal is processed using an audio card, such as when recording, processing or amplifying live music, the stereo microphone 21 outputs analog audio signals to the left side circuit 203 and the right side circuit 203a, respectively. Enter. Each individual signal is an analog
Received by digital converters (ADCs) 207 and 207a and converted to digitized audio signals.

In the preferred embodiment of the present invention, the contents of multimedia play file 201 are input to audio card 23 for processing. This play file can be a music, voice text, or multimedia
It can consist of other digital audio information that is part of the presentation. In particular, in the present invention,
We are considering an interactive speech education program that requires students to answer specific questions to supplement what they have learned from their teachers. The multimedia play file 201 is input and stored in the memories 205 and 205a.
These memories are RAMs and are used as buffers for storing the digital play file information before processing. Timers 211 and 211a
Exists to coordinate the loading of play file data from memory into the digital signal processors (DSPs) 219 and 219a, as well as the overall coordination of audio processing. Note that for ease of explanation, two digital signal processors are shown in FIG. 6, one for each channel. However, one of ordinary skill in the art will appreciate that a single DSP can provide processing functions for both left and right channels, and that single DSP method is also within the scope of the present invention.

Additional memories 213 and 213a are RA
It is M. These memories are loaded by the system to store the user profile data obtained in the hearing test by the system of the present invention and a set of instructions loaded into the DSP 219 to implement the present invention. Used for. Additional memory 213 and 213
a is a single memory, for example a chip, with separate loadable areas for profile data and instructions.
It may be a unit or two (or more) separate memories connected in parallel. Reference numbers 209 and 20
9a represents an interface capable of outputting audio data to be processed by the DSP and receiving input from a musical instrument or other audio source. In the present invention, a musical instrument digital interface (MIDI) may be used to provide the interfaces 209 and 209a.

Digital signal processors 219 and 2
19a is essentially a computationally intensive microprocessor such as the model TMS320C51 available from Texas Instruments Incorporated. These microprocessors must be relatively fast so that the play file can be modified in real time according to the profile data. That is, the play file data input to the DSP must be equal to the data output from the DSP. If they are not equal, there will be a delay and, in a multimedia environment, the user will see the output on the display screen but will not hear the corresponding audio data until a certain amount of time has elapsed. Note that in a multimedia environment, any kind of delay between video output and corresponding audio output is unacceptable.

The present invention can be implemented in software, which controls the operation of the DSP with instructions added to the set of instructions that control the normal operation of the DSP. The functions provided by these additional instructions include Fast Fourier Transform Functions (FFT) 221 and 221a, Play
There is a modification function for files 223 and 223a, and inverse Fourier transform operations 225 and 225a. These functions are shown in FIG. 6 in the order they are performed. In operation,
An instruction from memory 213 instructs the DSP to perform an FFT on the play file data. Basically, the FFT operation decomposes the audio data into component frequencies that make up a more complex signal and creates a list of frequencies (tones) present in the audio data of the play file. In addition, the FFT list contains the volume of each tone (Fig. 8). That is, the FFT is the play file 20.
1 provides a list of frequencies and magnitudes for each tone stored in memories 203 and 203a. Note that the FFT operation essentially translates the samples of the play file into frequency domain data. Other methods of converting the play file data into frequency components are also contemplated by the present invention. For example, a method using wavelets to extract frequency components has been shown to give good results.
This method is another preferred embodiment of the present invention.

Once the FFT for the play file data has been generated, the present invention modifies the data to compensate for the hearing impairment of the user (FIGS. 8 and 9). As mentioned above, the profile data in memory 213 contains the multiplication factor for each tone that is impaired by the user's hearing. The DSP then compensates for the hearing loss by increasing or decreasing the volume output at the particular frequency where the impairment is present. In other words, the volume of the tone in which the hearing impairment is present is amplified by an amount equal to the summing factor determined during the hearing test described above with respect to FIGS.

An inverse Fourier transform operation is then performed on the play file to invert the first FFT and return the data to the normal time-based format. The modified play data can now be output, and the voice level corresponding to the user's hearing impaired tone is increased or amplified by the DSP. The FFT operation 221, the correction function 223, and the inverse Fourier transform operation 225 are performed in real time. That is, all of these features operate on different tones in the play file simultaneously. Therefore, the user does not recognize that the voice output has been modified. However, the test subject will not be able to
File) can be heard more clearly.

The play file data has been modified,
After being converted again from the FFT list into the standard digital format, it is input to the digital / analog converters 215 and 215a. Digital-to-analog converter 215 converts the modified digital play file into an analog signal that is then made audible to the subject through headphones 27.

8 and 9 are flow charts representing the process used to modify the multimedia play file 201 in the present invention. As mentioned above,
This process is implemented in software and is performed just before the digital play file data is output to digital to analog converter 215. The correction process determines the frequency range of the output data and finds the multiplication factor from the user profile table. The hearing compensation process then modifies the audio in each audio range under test by the appropriate amount, as needed. As shown in FIG. 7 and described above, this process can operate on the left and right channels simultaneously. In general, this process either directly scales the input frequency data and then transforms it into a time domain for output, or bandpasses the input time domain signal to each test frequency range, and then passes the bandpass output to the subject profile table. And then offset or scaled according to, then recombined or mixed different frequency ranges for output.

In step 1, the process begins and in step 2 the user is identified. Next, in step 3, the user profile data and instructions are loaded from RAM 213 into DSP 219. Next, in step 4, FFT operation is performed on the play file data,
The volume at a particular frequency is retrieved. Step 5
At, the volume at a particular frequency is modified according to the subject profile data. For example, if the subject is 4.
In the audible frequency range from 0 KHz to 5.04 KHz,
It is assumed that it did not react until the volume reached 40 db.
The normal response of the user to this tone is that the volume is 10
It happens when it reaches db (this volume depends on the frequency of the tone). Therefore, the offset or addition coefficient in this range is 20db (30db-10db).
Multiplied by 31.6 (previously calculated), and the volume of this particular user in this particular range is based on the volume level of the normal user multiplied by 31.6. The frequency range in which the subject responds “normally”, that is, the user responds at a volume level that is statistically regarded as “normal”, is changed because the addition coefficient becomes zero (or the multiplication coefficient becomes 1). Instead, it passes through the audio card 23 and the circuits 203 and 203a.

In step 6, an inverse Fourier transform operation is performed on the play file containing the modified tones and the unmodified tones to which the user has responded normally to return the data to digital audio format. Next, step 7
Then, the DAC 215 converts the modified play file data from digital data to analog data. The hearing impairment compensation process of the present invention is essentially complete at this point and the modified play file in analog form is sent from the DAC 215 to the headphones 27 (step 8). The process then proceeds to step 9 and ends.

The invention described by the process shown in FIGS. 5, 6, 8 and 9 can be implemented in software and stored on a computer readable medium such as disk storage 20 of FIG. Please note. Those skilled in the art will appreciate that the disk storage device 20 may include a hard drive,
It will be appreciated that this includes tapes, floppy disk drives, etc. Furthermore, the present invention is also based on FIG.
The RAM 14 and the ROM 16 shown in FIG.

FIG. 10 is a graph representing the audio play file data after performing the FFT and before performing the modification according to the present invention. Specifically, the frequency range of tones in the play file is shown on the vertical axis (50
Hz, 200Hz, 1KHz, 10KHz, 15KH
z, a frequency of 20 KHz is illustrated), and the volume at each frequency is represented by a line extending to the right from the specific frequency. The length of the line is proportional to the volume, for example, the longer the line, the higher the volume level. In this example,
User hearing is 50Hz, 200Hz, 1KHz, 10
It is assumed that there is a failure at KHz and 15 KHz. Furthermore, as described above, the hearing test of the present invention has determined the addition coefficient corresponding to each of these frequencies. FIG. 11 is a similar graph after modifying the play file data according to the present invention. More specifically, as indicated by reference numeral "AF", the volume at the above-mentioned frequency is increased to compensate for the hearing loss of the subject. Following this compensation, the play file data is converted from the FFT format and output to the subject through headphones as described above.

While a few preferred embodiments have been shown and described, it will be appreciated that numerous changes can be made without departing from the scope of the invention.

The embodiments of the present invention will be described below. (1) determining a user profile based on the hearing ability of a user of the computer system; and the computer based on the user profile.
A method for modifying audio output of a computer system, comprising: modifying audio output of the system; and providing the modified audio output to the user. (2) The determining step gives the user at least one
Providing one tone, receiving input from the user when the at least one tone is at a volume level heard by the user, and the volume of the at least one tone until receiving input from the user. The method according to (1), comprising the step of: (3) further comprising saving the at least one tone at a volume level at which the user first heard the at least one tone, and offsetting based on the volume level at which the at least one tone was heard by the user. Creating the user profile, including calculating a factor, a plurality of tone ranges, a corresponding volume level when the at least one tone is heard by a user, and an offset factor corresponding to the plurality of tones. And a step of
The method according to (2). (4) The modifying step includes storing the user profile on the computer system, and storing at least one audio play file on the computer system.
Converting audio tones in the play file to a volume and corresponding frequency and adjusting the volume with the offset factor according to the user profile to create a play file containing the modified volume Characterized by including steps and,
The method according to (3). (5) the providing step includes converting the modified volume of the modified audio tone, and storing the modified audio tone in the play file to create a modified play file. And outputting the modified play file for the user. (6) The method of (5), wherein the step of determining a user profile comprises the step of testing the presence of headphones in the computer system. (7) Means for determining a user profile based on the hearing ability of the user of the computer system, means for modifying the audio output of the computer system based on the user profile, and the modified to the user. A system for modifying the audio output of a computer system, comprising: means for providing an audio output. (8) The determining means provides at least one tone to the user, means for receiving input from the user when the at least one tone is a volume level heard by the user, and from the user Means for incrementing the volume of the at least one tone until receiving an input of. (9) Further, based on means for saving the at least one tone at a volume level when the user first heard the at least one tone, and based on a volume level when the user first heard the at least one tone. Said user profile comprising means for calculating an offset factor, a plurality of tones, a corresponding volume level when said at least one tone is first heard by a user, and an offset factor corresponding to said plurality of tones. And means for creating,
The system according to (8). (10) The modifying means stores the user profile on the computer system, stores at least one audio play file on the computer system, and plays the play file.
Means for converting audio tones in the file to a volume and corresponding frequency; and means for adjusting the volume according to the user profile based on the offset factor to create a play file containing the modified volume. The system according to (9), further comprising: (11) The providing means stores the modified audio tone in the play file, and means for converting the modified volume of the modified audio tone,
The system according to (10), comprising means for creating a modified play file and means for outputting the modified play file for the user. (12) The means for determining a user profile comprises:
A means for testing whether or not headphones are present in the computer system;
The system described in 1). (13) Testing users of computer systems,
Means for determining a user profile based on the user's hearing, means for modifying the audio output of the computer system based on the user profile, and providing the modified audio output to the user A computer program stored on a computer-readable medium for controlling a computer system by modifying the audio output of the computer system. (14) means for causing the user to be provided with at least one tone, and means for receiving input from the user when the at least one tone is at a volume level audible to the user. , Means for incrementing the volume of the at least one tone until receiving input from the user. (15) Further, based on means for saving the at least one tone at a volume level when the user first heard the at least one tone, and based on a volume level when the user first heard the at least one tone. Said user profile comprising means for calculating an offset factor, a plurality of tones, a corresponding volume level when said at least one tone is first heard by a user, and an offset factor corresponding to said plurality of tones. And means for creating,
The computer program according to (14). (16) In the play file, the modifying means stores the user profile on the computer system, stores at least one audio play file on the computer system. Means for converting audio tones to a volume and corresponding frequency; and means for adjusting the volume by the offset factor according to the user profile to create a play file containing the modified volume. The computer program according to (15), which is characterized in that: (17) The means for causing the audio output to be provided to the user includes means for converting the modified volume of a modified audio tone, and storing the modified audio tone in the play file. The computer program according to (16), further comprising: means for instructing the user to create a modified play file, and means for outputting the modified play file to the user.

[0036]

Accordingly, the present invention is a particularly useful apparatus and method for compensating the output of a multimedia system based on the hearing impairment of a subject to help the user enjoy and understand the multimedia presentation. You will see that there is. The present invention is applicable to education, training, and many multimedia environments. The hearing test of the present invention is also extremely effective in determining whether school children have a hearing impairment that can interfere with learning.

[Brief description of drawings]

FIG. 1 is a block diagram of a computer system that can utilize the present invention.

FIG. 2 is a graph showing the relationship between the hearing frequency range and the sound volume at which the average person's sound volume is equal.

FIG. 3 is a graph similar to FIG. 2 of hearing for people with hearing impairment at approximately 200 Hz and 1 KHz.

FIG. 4 is a schematic diagram of a circuit for detecting the presence of a pair of headphones.

FIG. 5 is a flow chart representing a process for creating a user profile that stores all of a user's hearing impairments.

FIG. 6 is a flow chart representing a process for creating a user profile that stores all of a user's hearing impairments.

FIG. 7 is an audio card modified according to the present invention to compensate for hearing impairments present in a user profile.

FIG. 8 is a flow chart representing a process for modifying a multimedia audio file according to a user profile in the present invention.

FIG. 9 is a flowchart representing a process for modifying a multimedia audio file according to a user profile in the present invention.

FIG. 10 is a graph showing a user profile obtained from the results of a hearing test.

FIG. 11 is a graph showing a modified hearing profile after correcting a user's hearing impairment in accordance with the present invention.

[Explanation of symbols]

 21 Microphone 22 User Interface Adapter 23 Audio Card 27 Headphones 28 Speaker 100 Common Source Reference Voltage Circuit R7 Resistor C Capacitor 102 Comparison Mechanism 106 Large Time Constant RC Circuit 207 Analog-to-Digital Converter 201 Multimedia Play File 211 Timer 219 DSP 221 Fast Fourier transform mechanism 225 Inverse Fourier transform operation

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Claims (9)

[Claims] 1. A method of determining a user profile based on a hearing ability of a user of a computer system; modifying a voice output of the computer system based on the user profile; Providing a modified audio output, the method comprising modifying the audio output of a computer system. 2. The determining step includes providing at least one tone to the user; receiving input from the user when the at least one tone is a volume level heard by the user; Increasing the volume of the at least one tone until receiving input from a user. 3. The volume level at which at least one tone is first heard by a user.
Saving one tone, calculating an offset factor based on the volume level at which the at least one tone was heard by the user, multiple tone ranges, and the at least one tone heard by the user 3. The method of claim 2 including the step of creating the user profile that includes corresponding volume levels of time and offset factors corresponding to the plurality of tones. 4. A means for determining a user profile based on a hearing ability of a user of a computer system; a means for modifying an audio output of the computer system based on the user profile; Means for providing a modified audio output, the system for modifying the audio output of a computer system. 5. The determining means provides means for providing at least one tone to the user; means for receiving input from the user when the at least one tone is a volume level heard by the user; Means for incrementing the volume of the at least one tone until receiving input from a user. 6. The volume level at which at least one tone is first heard by a user.
Means for saving one tone, means for calculating an offset factor based on a volume level when the at least one tone is first heard by the user, a plurality of tones, and the at least one tone being heard by the user for the first time. 6. The system of claim 5, comprising means for creating the user profile that includes a corresponding volume level at which a tone is played and an offset factor corresponding to the plurality of tones. 7. A means for testing a user of a computer system to determine a user profile based on the user's hearing and a means for modifying the audio output of the computer system based on the user profile. And means for causing the user to be provided with the modified audio output.
A computer program stored on a computer-readable medium for controlling a computer system by modifying the audio output of the system. 8. The means for causing the user to provide at least one tone to the user; and the input from the user when the at least one tone is a volume level heard by the user. 8. A computer program as claimed in claim 7 comprising means and means for incrementing the volume of the at least one tone until receiving input from the user. 9. Further, said at least one tone at a volume level when first heard by a user.
Means for saving one tone, means for calculating an offset factor based on a volume level when the at least one tone is first heard by the user, a plurality of tones, and the at least one tone being heard by the user for the first time. 9. The computer program according to claim 8, further comprising: means for creating the user profile, which includes a corresponding volume level when the tone is played and an offset coefficient corresponding to the plurality of tones.