This invention relates generally to hearing correction devices and processes and more particularly concerns hearing normalization and correction for users with mild or moderate hearing loss.
Hearing loss can occur at any age and many people over the age of 18 begin to experience some deterioration in their hearing response. Tolerance of loss or deterioration may vary depending upon personal interests and occupations, with musicians being perhaps least tolerant. Those with hearing loss all share a common desire to restore their hearing to normal, considered to be a 0 dB threshold of hearing.
The parameters of normal hearing have been well documented and known for many years. But state-of-the-art hearing aids fall far short of the ability to restore the normal frequency range and intensity of hearing with any level of precision. Unfortunately, despite modern digital technology, design and performance remains generally directed to frequencies between 250 Hz and 8000 Hz and focused primarily on the frequency spectrum of speech.
The typical sound pressure level of speech increases in noisier environments like a restaurant or other public setting. And equal-loudness contours dating back almost a century, show that normal hearing has natural changes in frequency response based on changes in sound pressure levels. In reality, all sounds heard in the real world have harmonic structure associated with fundamental frequency components.
The typical adjustment method used for digital hearing aids is to do an initial tune based on an audiogram taken at the threshold of hearing. Then the audiologist will use the hearing aid testing program and plays with the software settings until the user is comfortable. In many cases numerous visits are required to get a comfortable and acceptable tuning.
But the measured threshold data made with pure tones does not accurately reflect low level hearing for a person with hearing impairment and the additional adjustments for higher levels are arbitrary guesswork. The end result does not represent a normal hearing response at any listening level.
The presently known best effort is an adaptive hearing aid which changes settings based on input sound pressure levels by dividing the audio spectrum into multiple, typically nine or more, frequency bands and then applying dynamic range compression in each band. There are numerous problems associated with this multiband compression approach.
First, compression threshold settings are selected based on a guesswork assumption as to when compression needs to start. Hearing response changes with spectral content and the data used for tuning is based on pure tones. Typical compressor threshold settings are at 65 db which means gain reduction will not start until the input sound pressure level is above 65 db. Incorrect frequency gain is the cause of common user complaints of unnatural sound and very poor fidelity. Second, the frequency relationship of phonetic spectral energy in speech is greatly altered. Serious audible artifacts are associated with both compressor gain overshoot, which occurs with sudden loud signals, and the following release time required to return to the low sound pressure levels gain setting. Prior art implementations have had to make a selection between fast acting compression and slow acting compression with a number of tradeoffs based on either selection. Third, the release time causes problems in hearing soft input signals that come immediately after loud input signals. Decreasing the release time in multiple frequency bands will result in increased distortion artifacts which will further reduce speech clarity and overall sound quality for the user.
In sum, known hearing aid systems take threshold measurements using only pure tones which do not accurately reflect how we hear real world sounds. They take actual measurements only at low level threshold of hearing. They are tuned by arbitrary listening. They have severe artifacts including overshoot, spectral modulation and poor release tracking. They produce unnatural and thin sound with poor overall fidelity. And they provide inadequate headroom due to low battery voltage.
It is, therefore, an object of the invention to provide a hearing normalization and correction system which affords hearing correction suited to the audio-quality demands of music industry professionals. Accordingly, it is also an object of the invention to provide a hearing normalization and correction system which affords accurate hearing correction. Yet another object of the invention is to provide a hearing normalization and correction system which affords an improved frequency response and dynamic range. And it is an object of the invention to provide a hearing normalization and correction system which accurately reflect real world sounds. It is another object of the invention to provide a hearing normalization and correction system which is capable of accurately measuring the actual hearing response of a user. A further object of the invention is to provide a hearing normalization and correction system with which users can self-test and measure their own actual hearing response. It is also an object of the invention to provide a hearing normalization and correction system which is capable of dynamic hearing correction without audible artifacts. And it is an object of the invention to provide a hearing normalization and correction system capable of converting and applying accurate measured data to enable automatic self-tuning of hearing correction responses.
In accordance with the invention, there is provided a hearing normalization and correction system for delivering to the ears of a user of an acoustical device an accurate hearing response to an input audio signal and for customized automatic tuning for the user of the device.
In the hearing normalization and correction process, the audio input signal is modified by a first correction response based on the actual hearing response of the user at a first sound pressure level to produce a first correction level response. The audio input signal is also modified by a second correction response based on the actual hearing response of the user at a second sound pressure level higher than the first to produce a second correction level response. The first correction level response is applied to the output signal of the acoustical device when the input sound pressure is at the first sound pressure level. The second correction level response is applied to the output signal of the acoustical device when the input sound pressure is at the second sound pressure level. When the input sound pressure is between the first and second sound pressure levels, the output signal is dynamically varied between the first and second correction level responses in correlation with the varying sound pressure level of the input audio signal. The second sound pressure level of the input audio signal may be the normal conversational speech level of hearing.
The audio input signal may also be modified by a third correction response based on the actual hearing response of the user at a third sound pressure level higher than the second to produce a third correction level response, in which case the third correction level response is applied to the output signal of the acoustical device when the input sound pressure is at the third sound pressure level and, when the input sound pressure is between the second and third sound pressure levels, the output signal may be dynamically varied between the second and third correction level responses in correlation with the varying sound pressure level of the input audio signal.
The audio input signal may be further modified by additional correction level responses based on the actual hearing responses of the user at additional corresponding sound pressure levels sequentially increasingly higher than the third to produce additional corresponding correction level responses. Each additional corresponding correction level response may then be applied to the output signal of the acoustical device when the input sound pressure is at the additional corresponding sound pressure level. When the input sound pressure is between additional corresponding sequential sound pressure levels, the output signal may be dynamically varied in correlation with the varying sound pressure level of the input audio signal.
Whatever the number of correction level and additional correction level responses may be, the audio spectrum may be divided into multiple frequency bands and the process repeated for each of the multiple frequency bands.
Modifying any correction response may be accomplished by measuring the corresponding actual hearing response of the user at the corresponding sound pressure level of the input audio signal and converting the measured actual hearing response into the corresponding correction level response.
In the hearing normalization and correction processor, a first converter receives the audio input signal and produces a digital output signal. A detector modifies the digital output signal to produce a control signal corresponding to the sound pressure level of the audio input signal. A first filter modifies the digital output signal to produce a first correction equalization signal corresponding to an actual measured first low level hearing response of the user. A second filter modifies the digital output signal to produce a second correction equalization signal corresponding to an actual measured second higher level hearing response of the user. A first multiplier dynamically varies the gain of the first correction equalization signal to provide a first maximum gain output signal when a corresponding detected sound pressure level is low. A second multiplier dynamically varies the gain of the second correction equalization signal to provide a second maximum gain output signal when a corresponding detected sound pressure level is high. A summer combines the first and second maximum gain output signals when the detected sound pressure level is between the high and low detected sound pressure levels.
In the tuning process, to provide an accurate low level hearing correction response, a shaped noise with a center frequency at critical frequency points is applied to an ear of a user to determine an actual low sound pressure level hearing response of the user. The determined actual low sound pressure level hearing response is converted into a low level correction response. The low level correction response is applied to the output of the acoustical device. To provide an accurate higher level hearing correction response, the tuning process further applies broadband masking noise at another sound pressure level higher than the low sound pressure level to the ear of the user and a narrow band stimulus with a center frequency at critical frequency points to the ear of the user to determine an actual higher level hearing response of the user. The determined actual higher level hearing response of the user is converted into a higher level correction response. The higher level correction response is then applied to the output of the acoustical device.
In the tuning process for providing an accurate speech level hearing correction response for a user, broadband masking noise at a speech sound pressure level is applied to the ear of the user. Narrow band stimulus with a center frequency at critical frequency points is applied to the ear of the user to determine an actual speech sound pressure level hearing response of the user. The determined actual speech sound pressure level hearing response of the user is converted into a speech sound pressure level hearing correction response. The speech sound pressure level hearing correction response is then applied to the output of the acoustical device.
In the automatic tuning process, a shaped noise with a center frequency at a set of selected frequency points is applied to the ear of the user to produce a shaped-noise set of frequency point data at an actual shaped-noise sound level hearing response of the user. A broadband masking noise is also applied to the ear of the user. A narrow band stimulus with a center frequency at a set of selected stimulus frequency points is applied to the ear of the user to produce a set of stimulus frequency point data at an actual stimulus level hearing response of the user. The shaped-noise and stimulus sets of frequency point data are stored in the memory of a digital processor. A command transmitted to the digital processor causes the digital processor to use the stored frequency point data to calculate noise level and stimulus level hearing correction responses and to use the noise level and stimulus level hearing correction responses to determine filter coefficients enabling the digital processor to provide accurate noise level and stimulus level hearing correction response curves.
Other objects and advantages of the invention will become apparent upon reading the following detailed description and upon reference to the drawings in which:
While the invention will be described in connection with preferred embodiments thereof, it will be understood that it is not intended to limit the invention to those embodiments or to the details of the construction or arrangement of parts illustrated in the accompanying drawings.
Hearing is the sensorial perception of sounds by the physiological mechanisms of the human ear. Sound input is perceived as pitch, loudness and direction based on its frequency and on its arrival-time difference to the ears. From this input we can detect musical quality, spatial information and even nuances of voiced emotion.
Pitch is the perception of frequency and is not greatly affected by other physical quantities such as intensity. Normal human hearing encompasses frequencies from 20 to 20,000 Hz. Spatial cues in sound typically come from the higher frequency information and in order to determine directivity require hearing this higher frequency information with both ears.
Loudness is the perception of intensity or sound pressure level. The ear is remarkably sensitive to low-intensity sounds. The lowest audible intensity, or threshold, is commonly referred to as 0 dB hearing level. Sounds as much as 1012 more intense can be briefly tolerated. At any given frequency, it is possible to discern differences of less than 1 dB and changes of 3 dB are very easily noticed.
Frequency does also have a major effect on perceived loudness. The ear has its maximum sensitivity to frequencies in the range of 2000 to 5000 Hz, so sounds in this range are perceived as being louder than, for example, those at 500 or 10,000 Hz, even if they all have the same intensity. And sounds near the high and low frequency extremes of the hearing range seem even less loud, because the ear is even less sensitive at those frequencies.
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Speech intelligibility is most critical for those with hearing impairment and has been the main focus of hearing aids for years.
Moving on to
The most sensitive frequency region of normal hearing at all sound pressure levels is approximately 3 khz. The audiogram of
Known multiband compression systems have severe artifacts including overshoot, spectral modulation and poor release tracking.
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The attack and release problems identified with respect to
In accordance with the invention, a hearing normalization and correction system is provided that delivers an accurate hearing response to the ear of a user of an acoustical device. The system relies on more meaningful and accurate measurement methods in order to provide the user with a dynamic response which can provide very natural sound.
Threshold measurements taken using only pure tones do not accurately reflect how real world sounds are heard. As discussed in relation to
In
Other stimuli, such as a tone cluster of multiple frequencies with the dominant frequency at the frequency of measurement, might be used with a broader spectrum of frequencies to produce a similar measurement result. In all cases the resolution of the measurement frequencies is critical to produce an accurate correction response. Such measurements when applied to a correction response will represent normalized hearing at a low sound pressure level.
In addition to low level measurement, a meaningful measurement at a higher sound pressure level is critical to delivering accurate and normalized hearing correction to the user. The second most desirable level for actual measurement is the sound pressure level of typical speech. Therefore, looking at
Other forms of masking, such as bandwidth limited noise centered at the measurement frequency or multiple masking tones near the measurement frequency, can also be used. Other stimuli than pure tones can also be used as long as the dominant frequency is at the measurement frequency of interest. The resulting collected measurement data will provide a real and accurate assessment of the actual measured hearing response at the higher sound pressure level. While the level of speech is considered to be the most common listening level, other sound pressure level measurements can be made if higher sound pressure resolution is desirable, as will be hereinafter discussed in relation to
The masked noise response MNR is the actual measured response using the higher sound pressure level measurement method described with respect to
Referring to
The system receives an input signal either from the input microphone M1 or via a direct wireless/Bluetooth interface WB from another transmitting device such as a cell phone or computer (not shown). The microphone M1 feeds a microphone preamplifier MP1 which feeds the input of an analog-to-digital converter ADC. The converter ADC provides a digital output signal to the processing core DSP1. In professional applications where increased headroom is critical, such as professional musical performances, the system may further include positive and negative adaptive rail control circuits PARC1 and NARC1 which operate to allow increased headroom for the microphone input signal if required to avoid clipping or overdriving the input microphone preamplifier MP1.
The output of the processing core DSP1 feeds a digital-to-analog converter DAC which provides an analog output signal to drive an output amplifier A1. The output of the amplifier A1 provides output voltage and current to deliver sound to a driver or acoustical device D. As described above, in professional applications where increased headroom is critical, such as professional musical performances, the system may further include positive and negative adaptive rail control circuits PARC2 and NARC2 which dynamically increase the output headroom of the system to avoid clipping the system. The control circuits PARC2 and NARC2 are identical in operation to PARC1 and NARC1 as described in reference to
The normalized hearing system may operate as a quality hearing normalization system with high precision hearing correction for professional audio applications but can also be used by any user in need of hearing correction. The hearing correction of the invention allows users with mild or moderate hearing loss anomalies to achieve a natural sounding response with both excellent frequency response and dynamic range.
The lower threshold SPL correction response curve 110 is based on measurements taken at 0 db SPL using the hereinbefore described shaped noise measurement method converted and applied as the required correction response curve with both higher bandwidth and higher resolution of testing. The higher correction response curve 120 is based on measurements taken at 60 db SPL using the hereinbefore described masking noise measurement method and reflects the correction response required to compensate for any measured hearing deficit. Other correction response curves at other measured sound pressure levels can also be applied if higher resolution testing is performed at additional SPL levels.
The adaptive hearing normalization system operates to dynamically vary between two or more measured response correction curves in correlation to the actual input sound pressure level that appears at the audio input of the hearing normalization system. Correlation relates to the direction of change in sound level and not to its absolute magnitude. Dynamically adaptive operation is required to provide the listener with the most natural sounding audio response and as close to normal hearing as possible. If the threshold of hearing for the listener produces more natural response at low sound pressure levels, the listener will feel as if normal low level hearing is restored.
By dynamically varying between multiple frequency responses at the correct SPL levels based on actual measured data, normal listening can be restored for a user with mild to moderate hearing loss. By increasing the number of response measurements by using masking noise at multiple higher SPL levels, an even more precise restoration of natural hearing response will be achieved for the user. Those with more severe loss will find great improvement when applying the additional higher level correction response curves.
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The digital output signal from the converter ADC 70 is applied to corrective filters 10 and 20 and to the SPL detector 30. One filter 10 applies corrective equalization based on the low sound pressure level measurements and the other filter 20 applies corrective equalization based on the higher 60 db SPL measurements. The outputs of the corrective filters 10 and 20 are applied to the inputs of multipliers 40 and 50, respectively. Even higher performance is possible by applying cross-fade in multiple frequency bands. The multiple correction curve filters can also be applied as multiple frequency band filters by dividing the audio spectrum into multiple frequency bands and applying the required low level and higher level gain at the required frequency points within in each frequency band. Separate level detectors and adaptive dynamics control would also be required for each frequency band.
The corrective filters 10 and 20 can be implemented with Infinite Impulse Response (IIR) or Finite Impulse Response (FIR) techniques. The filter coefficients can be calculated from the measured sound pressure level data using a number of methods documented in DSP literature, including Inverse FFT, fast convolution via FFT and Least Squares techniques.
The output levels of the multipliers 40 and 50 are controlled by the SPL detector 30 which provides a level control based on the actual sound pressure level of the audio at the audio input of the signal processor DSP1. When the SPL level at the audio input A1 is below 10 db SPL, the higher level multiplier 40 will be at a gain of 0 and the low level multiplier 50 will be at a gain of 1. As the input audio level increases above 10 db SPL the low level multiplier 50 will start to attenuate and the output of the higher level multiplier 40 will begin to increase. When the audio input level reaches 60 db SPL the low level multiplier 50 will be at a gain of 0 and the higher level multiplier 40 will be at a gain of 1. The outputs of the multiplier 40 and 50 are applied to the inputs of a summer 60. The output of the summer 60 is applied to the input of a digital-to-analog converter 80 which provides the audio output signal of signal processor DSP1. The audio output signal of the digital-to-analog convertor DAC 80 will be applied to an audio amplifier (not shown) which drives an acoustical device to provide sound to the ear of the user.
As hereinbefore discussed, additional correction filters can be added based on additional SPL measurement levels. If additional correction filters are used the SPL detector 30 will provide the control signal for the additional multipliers and cross-fade operation between the additional corrective filter outputs will be provided producing further enhanced operation.
The dynamic cross-fade operation between the corrective response curves can also be applied in multi-band operation with cross-fade in multiple bands between actual measured SPL levels. In the multiband approach, unlike prior art multiband compression systems, the actual required output level at different sound pressure levels of each band is applied based on actual measurements. Each frequency band would then dynamically vary or cross-fade between the two or more corrective response curves as determined by actual measurements. The multiband aspect of the invention does not use normal compression but rather a dynamic cross-fade. The dynamic cross-fade method can also actually provide improved speech articulation by increasing formant perception.
Turning to
The output signal VC of the slow release time constant filter 91 feeds a comparator 93 and a subtractor 95. The output signal VC is also a first output control voltage of the SPL detector 30. The subtractor 95 performs a mathematical function producing an output signal 1-VC to provide a second output signal of the SPL detector 30. The two control signals are applied to the gain multipliers 40 and 50 seen in
Returning to
The output signal S93 of the comparator 93 feeds the input of a negative peak control 94 and the output signal S94 of negative peak control 94 feeds the slow-release-time constant filter 91. In operation, the fast response rectifier filter 31 determines the maximum attack time of the SPL detector 30 and feeds both fast and slow time constant filters 90 and 91, respectively. A sudden loud audio input signal will produce a fast attack response as shown in
If the input signal drops quickly and over a large decibel range, the output control signals VC and 1-VC will provide a release response equal to the fast release time constant filter 90. If the input audio signal drops extremely slow, the difference between the slow release response filter output signal VC and the decibel release window output signal S92 will never exceed the 6 db window, so the slow release response VC will remain as the output response VC. This ensures that the slow decaying audio input signal will be processed by the slow release response and maintain ripple free release without any gain modulation during the cross-fade operation. Input audio signals with a moderately fast decaying envelope will produce an output time constant that tracks the actual envelope of the input audio. The tracking is due to the interaction of the decibel release window 92, the comparator 93 and the negative peak control 94 to increase the negative peak of the slow release time constant filter 91.
Due to the operation of the decibel release window 92, once the slow time constant negative peak is equal to the fast time constant, the slow release time constant becomes dominant. Therefore, only negative going peaks and not control signal ripple in the fast release time constant will affect the slow release, eliminating ripple that would otherwise occur in the control signal VC. This allows extremely fast release response without the associated gain modulation.
The release time required to return to the low level correction response is adaptive and will be based on the short term envelope of the audio input signal. If the sound pressure level drops quickly, the release response will track the input audio's envelope. The fast release signal S90 at the output of the fast release time constant filter 90 is provides a release time as fast as 3 ms. The slow release signal VC at the output of the slow release time constant filter 91 can be as much as 500 ms or more. The adaptive dynamics control will track the actual envelope of the incoming audio signal and provide an adaptive ripple free, smooth release response that avoids gain modulation that causes pumping and breathing artifacts in the processed audio. This is especially helpful when the user is in a loud environment where the sound pressure levels are changing quickly.
The release response can adapt over a ratio greater than 150:1 compared to the 8:1 adaptive release response of multiband compression systems. The dynamic response combined with the cross-fade operation affords an extremely adaptive and transparent hearing normalization system.
Looking at
Once the measurement data is collected at the various sound pressure levels and stored in memory, the automatic tuning operation will be available to the user. Selecting the automatic tuning operation will initiate a process whereby the cell phone, computer or DSP processor will use the stored measured response data to calculate and determine proper filter coefficients required to produce the multiple correction response curves. The filter coefficients are applied within the DSP processor to produce accurate correction response curves at the multiple sound pressure levels as illustrated in
Typical hearing aid and personal listening devices operate on batteries. Operating at lower voltages could increase operating time and the current available to power the device, but the lower the voltage the less the available voltage swing and headroom for both the input signal and to drive the output speaker. A response even close to that of a person with normal hearing requires a hearing normalization system operating with high dynamic range.
Look now at
Looking again at
The emitter of a transistor Q5 is connected to ground and a resistor R12 is connected between the −1.5 volt power supply rail and the collector of the transistor Q5. A resistor R10 is connected to the −1.5 volt power supply rail and the base of the transistor Q5. Another resistor R9 is connected between the base of the transistor Q5 and the cathode side of a diode D12. The anode side of the diode D12 is connected to the output of the microphone preamplifier U1. The values of resistors R9 and R10 are selected to bias the switching transistor Q5 on when the output of the microphone preamplifier U1 is below positive 0.3 volts. When the switching transistor Q5 is switched on, the collector of the switching transistor Q5 will be at ground. When the collector of the switching transistor Q5 is switched to ground, the transistor Qb is switched on, connecting the negative side of the capacitor C2 to the −1.5 voltage rail. This will charge the capacitor C2 across the +1.5 volt power supply rail and the −1.5 volt power supply rail. Therefore, the capacitor C2 will be charged to 3 volts. When the output of the microphone preamplifier U1 swings positive by more than 0.3 volts, the switching transistor Q5 turns off and the base of the transistor Q6 will be pulled to the −1.5 volt rail through the base resistor R11 and the resistor R12, switching off the transistor Q6, so the transistor Q6 is now open collector.
As the output of the microphone preamplifier U1 swings positive by more than 0.4 volts, a rail boost transistor Q4 becomes active. The collector of the rail boost transistor Q4 is connected to the +1.5 volt power supply rail. The emitter of the rail boost transistor Q4 is connected to the negative side of the capacitor C2. The base of the rail boost transistor Q4 is connected to the output of the microphone preamplifier U1 through series connected diodes D8, D9, D10 and D11 with the cathode side of the diode D11 connected to the base of the rail boost transistor Q4 and the anode side of the diode D8 connected to the output of the microphone preamplifier U1. The rail boost transistor Q4 operates as an emitter follower with a negative offset based on the forward diode drop of the diodes D8, D9, D10 and D11 plus the VBE drop of the rail boost transistor Q4. As the output of the microphone preamplifier U1 increases above approximately positive 0.4 volts the emitter voltage of the rail boost transistor Q4 starts to increase linearly above the −1.5 volt power supply rail to which the negative side of the capacitor C2 has been charged. This increases the voltage on the negative side of the capacitor C2 which then increases the voltage at the positive power supply pin of the microphone preamplifier U1. This voltage increase will track the audio input signal and continue until the output of the microphone preamplifier U1 exceeds 4 volts. The output will saturate at approximately 4.2 volts. This allows the output of the microphone preamplifier U1 to swing between positive 4.2 volts and negative 4.2 volts when the negative boost rail 150 operates.
This provides headroom for the microphone preamplifier U1 equal to that of an 8.4 volt battery, well above what would be normal with a 3 volt battery. This also provides an increase of nearly 3 times the available output voltage swing before clipping. The increased positive voltage is available due to the charge held in the capacitor C2. The circuit operates like a charge pump circuit controlled by the audio output signal. A slight voltage drop will result from the current pulled by the operation of the microphone preamplifier U1. This slight discharge will be replenished as the output voltage swing of the microphone preamplifier U1 drops below 0.3 volts, thereby turning on the positive rail charge circuit 40. A capacitor C1 and the capacitor C2 are selected to provide minimal discharge at very low frequency operation in order to avoid voltage sag of the +VAR peak voltage. Without the dynamic operation of the microphone preamplifier U1, the normal output swing would be +/−1.5 volts for a total voltage swing of 3 volts. The forward voltage drop of the shottkey diodes D1 and D2 becomes a limiting factor at lower voltages. A full 3 times increase in headroom would be possible with ideal diodes for D1 and D2. If very low battery voltage is used, critical selection of the diodes D1 and D2 is required to provide the lowest possible forward voltage drop. Referring to
Returning to
The emitter of the transistor Q2 is connected to ground and a resistor R7 is connected between the +1.5 volt power supply rail and the collector of the Q2. A resistor R5 is connected to the +1.5 volt power supply rail and the base of the transistor Q2. A resistor R6 is connected between the base of the transistor Q2 and the anode of a diode D7. The cathode of the diode D7 is connected to the output of the microphone preamplifier U1. The value of resistors R5 and R6 are selected to bias the switching transistor Q2 on when the output of U1 is above negative 0.3 volts. When switching transistor Q2 is switched on, the collector of the transistor Q2 will be at ground. When the collector of the transistor Q2 is switched to ground a transistor Q3 is switched on, connecting the positive side of the capacitor C1 to the +1.5 voltage rail. This will charge the capacitor C1 across the +1.5 volt power supply rail and the −1.5 volt power supply rail. This means that the capacitor C1 will now be charged to 3 volts. When the output of U1 swings negative by more than 0.3 volts, the switching transistor Q2 turns off and the base of the transistor Q3, through a base resistor R8, will be pulled to the +1.5 volt rail by another resistor R7, switching off the transistor Q3, so the transistor Q3 is now open collector.
As the output of the microphone preamplifier U1 swings negative by more than .4 volts, the rail boost transistor Q1 becomes active. The collector of Q1 is connected to −1.5 volt power supply rail, the emitter of Q1 is connected to the positive side of the capacitor C1 and the base of Q1 is connected to the output of the microphone preamplifier U1 through series connected diodes D3, D4, D5 and D6 with the anode side of the diode D3 connected to the base of transistor Q1 and the cathode of the diode D6 connected to the output of the microphone preamplifier U1. The transistor Q1 operates as an emitter follower with a positive offset based on the forward diode drop of diodes D3, D4, D5 and D6 plus the VBE drop of the transistor Q1. As the output of the microphone preamplifier U1 decreases below approximately negative 0.4 volts, the emitter voltage of transistor Q1 starts to decrease linearly below the +1.5 volt power supply rail to which the positive side of the capacitor C1 has been charged. This increases the voltage on the positive side of the capacitor C1 which then increases the negative voltage at the negative power supply pin −V of the microphone preamplifier U1. This negative voltage increase will track the audio input signal and continue until the output of the microphone preamplifier U1 exceeds −4 volts. The output will saturate at approximately −4.2 volts. The increased voltage is available due to the charge held in the capacitor C1.
As noted above, the circuit operates like a charge pump circuit controlled by the audio output signal. A slight voltage drop of the capacitor C1 will result from the current pulled by the operation of the microphone preamplifier U1. This slight discharge will be replenished as the output voltage swing of the microphone preamplifier U1 goes above negative 0.3 volts, thereby turning on negative rail charge circuit 20. The capacitors C1 and C2 are selected to provide minimal discharge at very low frequency operation in order to avoid voltage sag of the +VAR peak voltage. Without the dynamic operation of the microphone preamplifier U1, the normal output swing would be +/−1.5 volts for a total voltage swing of 3 volts. Referring again to
Also critical for normalized hearing is the dynamic range of the amplifier driving the acoustical device, which delivers sound to the ear. This becomes critical for the professional musician since stage sound pressure levels can be quite high and the output level required will be higher than nominal listening levels. There will also be times where the non-musician user may require higher output levels without distortion, especially critical to handle transients without clipping. While one of the available user selectable audio functions will be compression or limiting, thereby allowing the user to reduce the output level in loud environments, the ability to handle momentary loud levels is critical for providing a normal hearing response. The same method of increasing the available headroom for the microphone preamplifier is also used to increase the headroom of the output amplifier.
The dynamic headroom circuitry shown in
Alternatively, the dynamic correction response could be achieved using multiple frequency filters be implemented as either state variable filters or fixed bandwidth filters, similar to a graphic equalizer, dynamically varying the output level of the filters in correlation to the input sound pressure level to produce the required correction at the different sound pressure levels. This could be implemented in either analog or digital form.
Thus, it is apparent that there has been provided, in accordance with the invention, a hearing normalization and correction system that fully satisfies the objects, aims and advantages set forth above. While the invention has been described in conjunction with specific embodiments thereof, it is evident that many alternatives, modifications and variations will be apparent to those skilled in the art and in light of the foregoing description. Accordingly, it is intended to embrace all such alternatives, modifications and additions as fall within the spirit of the appended claims.
Filing Document | Filing Date | Country | Kind |
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PCT/US2020/054262 | 10/5/2020 | WO |
Number | Date | Country | |
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62909902 | Oct 2019 | US |