JPH0732505B2 - Headphone device - Google Patents

Description

Detailed Description of the Invention

[0001]

FIELD OF THE INVENTION The present invention relates generally to headphones, and more particularly to a novel method for reducing distortion while reducing noise while providing a relatively constant frequency response that is perceptible to the user. Equipment and technology. The present invention achieves the above features with a relatively compact headset that can be comfortably worn without exerting undue pressure on the head due to the force of pressing the cup against the head.

[0002]

2. Description of the Prior Art Typical conventional methods of damping noise use headphones with high mass, large volume, and spring support devices that exert a large pressure on the head. High mass increases inertia which resists acceleration and contributes to the structural rigidity of the headphones wall. High pressure has the effect of increasing the closed low frequency damping without air leakage. The large volume of compliant air cavity provides high frequency roll-off. However, most of these techniques increase user discomfort.

Conventional active noise cancellation techniques include methods of converting external noise using a microphone external to the headphones. The electrical device then processes the converted noise signal in the same way as the attenuation caused by the headphones on the noise signal and supplies the headphone driver with the opposite phased signal to cancel external noise. To do. This method is an open loop system that is not applicable to different users and may actually increase the noise level inside the headphones. Another method uses a closed loop or servomechanism, such as National Techni.
4 1975, described in report AB-A009274, distributed by cal Information Services.
“A STU” by Patrick Michael Dallosta of the Moon
DY OF PROPOSED EAR PROTEC
TION DEVICES FOR LOW FREQ
There is "UNCY NOISE ATTENUATION". U.S. Pat. No. 3,009,991 discloses a speed sensitive microphone located in the feedback loop, in the immediate vicinity of the speaker diaphragm. U.S. Pat. No. 3,562,429 discloses a movable feedback, a remote acoustic feedback and a feedback around the headphones.

[0004]

SUMMARY OF THE INVENTION Therefore, it is an object of the present invention to provide a headphone device which solves the above problems.

[0005]

According to the present invention, there is provided a means for defining a headphone cavity and an electroacoustic transducing means, such as a pressure sensitive microphone, provided within the cavity, so as to accommodate external noise in the same cavity. It provides a signal corresponding to the sum of the sounds produced by the headphone driver. It also has means for combining the converted signal with the desired input signal to be reproduced, an error signal representative of noise and the difference between the input signal to be reproduced and the output of the headphone driver in the cavity. To occur. Servo means, including combining means, compensate for these error signals to provide the ear with an output acoustic signal with significantly reduced external noise and distortion, and
It provides a nearly uniform frequency response between the input and the ear to which the desired signal to be reproduced is applied.

Cushion means are provided between the ears and the headphones to prevent significant leakage and prevent appreciable divergence in the pressure field. The cushioning means are sufficiently flexible to accommodate the irregular ear shape to provide the desired seal, have sufficiently high density and flow resistance, and the cavity portion defined has rigid walls. Acting as a near-ideal cavity, the pressure wave at the cavity wall satisfies the zero velocity boundary condition and the pressure amplitude is nearly constant for wavelengths well above the maximum distance across the cavity. The electro-acoustic transducing means is positioned proximate to the headphone driver means and responsive to the pressure in the cavity near the ear. Preferably, the headphone driver means comprises a diaphragm which acts as a piston in the relevant audio frequency range. The span across the diaphragm is desirably small, eg its diameter in a circular diaphragm, eg less than 1/3 wavelength at the highest frequency of the operating audio frequency.

[0007]

EXAMPLES The present invention will be described in detail with reference to examples.

Referring to FIG. 1, there is shown an outline of a state in which the headphone of the present invention is put on the ear. Microphone 11
Is arranged coaxially with the headphone housing 13, the driver 17, and the driver diaphragm 14 in the cavity 12, and the outer ear 16 and the cushion support portion 21 of the headphone 21.
A cushion 15 seals a region between the and. The microphone 11 is located near the entrance of the ear canal 18 and
The pressure wave at 1 is arranged so that the amplitude of the pressure wave is substantially the same as the entrance to the ear canal 18. The cavity 12 is reduced so that the pressure is substantially constant throughout the cavity.
To achieve this purpose, the cushion 15 has high mechanical compliance, high flow resistance and high density,
The axial cross-sectional area is approximately equal to the diaphragm 14 and smaller than the axial annular cross-sectional area of the cushion 15 around the cavity 12. The headphone housing 13 is joined to a resilient headband 23 in a known manner by a friction pole joint 22, as shown in FIG.

A typical material for the headphone cushion 15 is an open cell that is slow to recover.
It is a urethane foam. Cushion 15
Is used to hold the outer ear 16 over a relatively large area, distribute and effectively seal the force necessary to maintain a good seal for a sufficiently large area, and sufficiently reduce the pressure on the ear. I try not to make people uncomfortable. The open-cell high-flow resistance material exhibits the mechanical advantages of the open-cell material that conforms to the irregular shape of the ear, and at the same time, it significantly attenuates the spectrum component at a predetermined frequency in the middle range, for example, 2 kHz or more. It has the acoustical advantages of (closed cell) materials. It also keeps the pressure in the cavity substantially constant in this frequency range. Fluid-filled cushions also have these properties. Such an arrangement of the present invention provides the overhead power generated in response to the forces required to hold a closed or continuous bubble low flow resistance cushion around the ear that attenuates low frequency signals as is conventional. This is different from providing a large cavity with high pressure. Such conventional cavities are characterized by a divergence in a large pressure field and require a larger diaphragm displacement than the smaller cavities of the present invention to produce a constant sound pressure level. In this way, better acoustic efficiency can be achieved with a small set that gives the user of the present invention more comfort.

Referring to FIG. 2, there is shown a block diagram showing a logical configuration of the device according to the present invention. The signal combiner 30 algebraically combines the desired signal to be reproduced by the headphones at the input terminal 24 with the feedback signal provided by the microphone preamplifier 35. The signal combiner 30 supplies the combined signal to the compressor 31, which limits the level of the high level signal. Then, this compressor sends the compressed signal to the compensator 31A. Compensation circuit 31A adapts the open loop gain to the Nyquist stability criterion to prevent oscillation when the loop is closed. The device shown is similarly constructed for the left and right ears.

The power amplifier 32 energizes the headphone driver 17 to generate an acoustic signal in the cavity 12, which signal at the portion indicated by circle 36 and the external noise signal entering the cavity from the area of the acoustic input terminal 25. Combined, the combined sound pressure signal is applied to the microphone 11 where it is converted.
The microphone preamplifier 35 amplifies the converted signal and transmits it to the signal combiner 30.

The compensation circuit 31A has a loop gain T
(S) is 40 to 2000H as shown by the curve 20 in FIG.
Designed to be maximal in z (open loop).
This loop gain is represented by P (s) = (CBDEMA), where DM is the transfer function of the electrical signal output of the microphone 11 with respect to the electrical input to the driver 17, A,
B, C, E, D, and M are the microphone preamplifier 35, the power amplifier 32, the compressor circuit 31, and the compensation circuit 3, respectively.
1A is a transfer characteristic of the driver 17 and the microphone 11. The loop gain is maximized with consideration of the phase and amplitude margins sufficient to ensure stability under different conditions, including user head differences and decoupling.

The closed loop transfer function P _O from the electrical input to the pressure output is P _O / V _I = T _u = CBDE / (1 + CBDEMA) at the entrance to the ear canal. The amplitude of this closed loop transfer function as a function of frequency corresponds to curve 21 in FIG. The amount of active noise reduction when P _I corresponds to an acoustic noise input is (P _O / P _I ) = N _R = 1 + CBDEMA = 1 + T (s).

Referring to FIG. 3, a curve 23 representing the actual noise reduction measured by the microphone simulating the middle ear is shown in comparison with the theoretical curve 24 obtained by adding 1 to the open loop gain. .

Referring to FIGS. 5 and 6, there are shown block diagrams showing the theoretical construction of the preferred embodiment of the present invention. For the sake of convenience, the compensation circuit block 2 is represented by six blocks indicated by numerals 1 to 6, and the compensation circuit block 2 is further divided into sub blocks 2a, 2b, and 2c.
The compressor block 5 is further divided into 5 sub-blocks 5
It is divided into a, 5b, 5c, 5d and 5e. 5 and 6
The circuit portion forming the block of FIG. 7 is shown by a broken line boundary in FIGS. Those skilled in the art will be able to practice the invention by assembling the circuits of FIGS. 7 and 8 and need not be detailed. Next, referring to FIG. 5, FIG. 6 and FIG.
An embodiment of the present invention will be described with reference to FIG.

The adder / multiplier 1 has an input terminal 24.
And an adder 30 for receiving the input audio signal from the amplifier and the feedback signal provided from the amplifier 35. The adder 30 is realized by an operational amplifier connected in the form of a normal inverting addition amplifier, as shown in FIGS.
A capacitor connected between the connection point of a pair of similar input resistors and system ground shunts high frequency signal components to system ground. Analog switch U3
Half of 04 transmits the remaining low frequency components to the virtual ground at the input of integrated circuit U102. The modulated signal on the MOD line from compressor block 5 turns the switch on and off at a rate of 50 KHz. Multiplication is performed by controlling the switch closing time in each cycle. For signals characterized by spectral components with bandwidths much smaller than 50 KHz, the switch can be thought of as an impedance whose magnitude is proportional to the duty cycle of the modulating signal waveform. In normal operation,
The switch is closed most of the time. When the compressor 5 detects a very large input amplitude, the duty cycle is reduced, attenuating the low frequency spectral components of that input signal.

The resistors and capacitors connected in series directly transmit the high frequency signal component to the input of U102 so that they are not affected by the multiplication operation.

The output of the adder / multiplier block 1 is fed to a compensation block 2, which is characterized by amplitude and phase characteristics which guarantee the stability of the feedback loop without significantly reducing the overall loop gain. It consists of an active filter attached. Section 2c provides gain at high frequencies with proper roll-off. Sections 2a and 2b compensate for the loop gain phase response at low and high frequency crossover points, respectively. The principle of the preferred form of this compensation is described below, but it provides high gain in the frequency band containing most of the spectral components of the voice while maintaining stability preventing oscillations.

Power amplifier block 3 receives the output signal from compensation block 2. Section 3b is a well known non-inverting amplifier with a separate output current buffer. Category 3a is
A simple diode limiter protects the driver from destructive power dissipation levels. The light emitting diode emits light when the limiter operation is occurring. The input is preferably AC coupled to remove the DC offset from the previous stage.

The driver / microphone / ear system block 6 is not shown in FIGS. Driver 1
7 receives the amplified signal from the power amplifier block 3,
It produces an acoustic signal that is sensed by the ear 16 and converted by the microphone 11. Incomplete sealing of cushion 15 causes low frequency roll-off. Several KHz
The composite structure that causes multiple resonances at frequencies above
It is also a feature of block 6. In addition, the propagation delay from driver 17 to microphone 11 and the distributed source characteristics of driver 17 cause excessive phase shift. However, the components of the system work together to compensate for these non-uniform characteristics and produce a substantially uniform closed loop frequency response between the input terminal 24 and the ear canal 18.

Microphone preamplifier block 4
Is a low noise operational amplifier connected so as to have a non-inverting gain, which receives the converted signal from the microphone 11. The amplifier and gain are chosen to dominate the microphone's self-noise, thereby minimizing the effect of system electronics on the noise level at ear 16. The Zener diode supplies a bias voltage V _CC for the electret microphone 11. The amplifier 35 of the adder / multiplier block 1 receives the amplified signal provided by the microphone preamplifier 4.

The compressor block 5 monitors both the signal at the input terminal 24 and the feedback signal at the output of the microphone preamplifier 4 and modifies the modulated signal for modulating the low frequency gain of the adder / multiplier block 1 in the MOD line. Supply to. Section 5a sums the feedback signal and the input signal for both the left and right channels. A low pass filter having a corner frequency (typically 400 Hz) selectively transmits the combined signal for full wave rectification. Section 5b provides a rectified signal with a fast attack time and a slow decay.
y) provide an output signal that is averaged over time and proportional to the low frequency spectral energy of both the left and right loops. Section 5c converts the latter signal into a proportional current with an offset whose gain and offset are controlled by a potentiometer.

Section 5d receives the output current signal from section 5c and provides a 50 KHz modulated signal on the MOD line. Integrated circuit U305 of FIG. 7 comprises a 50 KHz clock pulse source that triggers integrated circuit U306 and resets its output to ground every 20 microseconds. Integrated circuit U3
The voltage drops linearly until it reaches a threshold level at a rate proportional to the output current provided by pin 2 capacitor voltage distribution 5c on 06, switching the output switch of integrated circuit U306 to high level and resetting the capacitor voltage on terminal 2. Keep positive power supply voltage until triggered again. Analog pin of adder / multiplier block 1 is control pin 1
And 8 for the ground potential, the adder / multiplier gain for low frequencies is inversely proportional to the current level provided by section 5c. The large current causes the capacitor potential on pin 2 of integrated circuit U306 to reach the threshold level quickly, and the analog switch U304 is correspondingly closed in a short time. Section 5e drives an LED bar graph display that displays the amount of compression. Since the low frequency spectral components carry most of the typical input signal, it is sufficient to sense the low frequencies and operate accordingly.

In summary, the system can be thought of as a servo system with two input signals. The first input is the audio electrical signal to be reproduced.
The second input is the acoustic noise signal around the ear. The output of this system is the acoustic signal generated in the ear. The feedback signal is a voltage proportional to the instantaneous sound pressure at the entrance to the ear canal. This sound pressure is a combination of the sound provided by the driver and the ambient acoustic noise. A small electret microphone 11 converts this signal, a preamplifier block 1 amplifies it, and an adder / multiplier block 1 adds this feedback signal with the input audio signal to give the actual sound pressure to the ear. An error signal is provided which represents the difference from the desired sound pressure. The desired sound pressure is proportional to the input audio signal. The compensation block 2 selectively transmits the spectral component of the error signal to ensure the stability of the loop. The amplifier block 3 amplifies the compensated signal and sends it to the driver 17 to generate a sound pressure at the ear corresponding to the desired audio input signal. Thus, over the frequency range in which the feedback loop is active, the loop is the spectral coloration of the driver / microphone / ear system.
The ambient noise is canceled by performing the correction for n).
The amount of correction is related to the amount of stable gain that the loop can deliver. The compressor block 5 cooperates with the multiplier portion of the adder / multiplier block 1 to prevent the input audio or acoustic signal from driving too far into the loop and clipping.

Having described the preferred embodiment of this system, the subsystem and its features will now be described. The compressor block 5 is used for compression artifacts.
It is implemented in a special beneficial way to reduce the fact) and avoid non-linear oscillations. Conventional compressors are typically classified into basic types, n-to-1 compression and threshold compression. The n: 1 compressor has an output level of 1 dB for each ndB input level.
Cause changes. It is linear with respect to an input signal below a certain threshold in threshold compression, and when it exceeds this threshold, it becomes an n: 1 compression ratio, and when this ratio is above the threshold level, it may become infinite and limit the average output level.

The n-to-1 compressor compresses a signal transmitted through a noisy communication channel or a signal to be recorded on a noisy medium, and then expands the compressed signal after detection to record or communication channel noise. Is widely used in companders that restore their original dynamic range with significant attenuation after expansion. Threshold compressors are commonly used in systems where the signal is not decompressed later because it has fewer unwanted artifacts in the output signal than n-to-1 compressors when properly designed.

A threshold compressor having an infinite compression ratio above the threshold is typically made with a feedback loop. If the compressor gain and compressor attack and decay time constants are not carefully selected, a significantly undesired audible sound may be produced, especially for input levels just above the threshold, and the system may oscillate. Produces an unpleasant audible sound.

The present invention is an improvement over conventional threshold compressors that have an infinite compression ratio above the threshold. FIG. 9 is a block diagram showing the logical configuration of the device implemented in block 5. This device responds to an input signal X at terminal 51,
The compressed signal Y is supplied to the terminal 52. The dividend input of the divider 53 is connected to the input terminal 51. Input terminal 51
Is also connected to the input of full wave rectifier 54. The output of full-wave rectifier 54 is connected to the input of averaging lowpass filter 55, which is characterized by a decay response time constant that is much greater than the attack response time constant. The output (X) of the averaging low pass filter 55 is connected to the input of an amplifier 56 having a compressor gain K.
Adder 57 receives signal K ₀ at one of its inputs and provides a divisor signal to the divisor input of divider 53. Divider 53 provides the quotient signal X / a to the input of output amplifier 58 which provides the compressed output signal Y.

For a static signal, eg a sine wave, the input-output gain is Y / X = 1 / (K ₀ + K (X))
It turns out that FIG. 10 is a graph showing this compression characteristic. This characteristic is Y / X = for small signals.
1 / K ₀ = constant, Y / X = 1 / for large signals
It is similar to the characteristics of a threshold compressor with an infinite compression ratio above the threshold, where KX or Y = 1 / K. However, the compressor according to the present invention has at least 2 compared to conventional compressors.
There are two advantages. Since the transition from the uncompressed state to the complete compression is performed smoothly, there are few audible compression generated sounds for a complicated input signal, for example, music. Furthermore, since there is no feedback, non-linear oscillation cannot occur.

Here, a preferable compression mode will be described.
If the attenuation of the gain A (ω) is known over the entire frequency range, then the phase φ (ω) for the minimum phase network
Is uniquely determined, and similarly, if φ (ω) is known over the entire frequency range, then A (ω) is for a “minimum phase” function with no poles and zeros in the right half of the s or p plane. It can be uniquely determined. This property, which is already known, is described in M.M. I. T. RADIATION LABORA
TORY SERIES vol. 25 "Theory
4.9 of of Servomechanisms "
"ATTENUTION PHASE RELATI
ONSHIPS FOR SERVOTRANSFER
FUNCTIONS ". This relationship was initially the Journal of Mathematic
cs and Physics, Y. C., June 1932. W. "NETWORK ANALYSIS AND FEE" by Bode, reported in Lee's paper.
DBACK AMPLIFIER DESIGN "
(D. Van Nostrand Co. New York, 1945), Chapter XIV, "Relations be."
tween Real and Imagery
Components of Network Fun
ctins ”. The above "Theor
y of Servomechanisms ".
In "8", the "damping-phase" type analysis is described as the most satisfactory approach to the servo design problem, and the criterion of the phase margin at the feedback cutoff frequency is a good practical criterion of the system stability. It is stated that it should be at least 30 °, preferably above 45 °. Above the cutoff, the gain A is 1 (log
It is stated that the frequency where A = 0) should be at least 2 1/2 octaves above the cut-off frequency to provide sufficient phase margin.

The purpose of providing the phase margin is to avoid a situation in which undesired oscillation can be continued and to eliminate peaking that amplifies external noise. Frequency f with gain of 1
_The disadvantage of providing such a large region between _c and the corner frequency is that the desired effect of negative feedback on spectral components within that frequency range is significantly reduced.
The present invention provides this shortcoming by providing an appropriate phase margin at unity gain frequency f _c by combining networks with high open loop gain while maintaining stability in the relevant frequency bands. In this way, both damping or gain and phase characteristics are established.

The present invention establishes a stable phase margin at frequency f _c where the gain drops to zero, while at the same time providing an open loop gain or damped frequency response with arbitrary slope at the pass band or edges. It includes a characterized compensation means. These principles are further understood by the examples described below.

Referring to FIGS. 11 (a) and 11 (b), there are shown graphs of gain or attenuation and phase characteristics of an operational amplifier having a normal first-order characteristic. The gain is constant up to half power or the corner frequency ω ₀ , and then linearly decreases at 6 db per octave. 12 (a) and 12
Referring to (b), the attenuation and phase characteristics shown in FIGS. 11 (a) and 11 (b) are modified by applying the principle of the present invention, and are improved at f ₀ or more while maintaining substantially the same phase margin. A graph is shown that achieves large loop gain.
This is accomplished by adding the portion indicated by the dashed line to the break point at f ₁ above f ₀ and providing a 12 db slope per octave. The modified phase characteristic shown by the broken line in FIG. 12B has a phase margin of approximately π / 2.

Referring to FIG. 13, there is shown a modified bandpass response with dashed lines representing the gain or attenuation modifications to the more conventional approach with a slope of 6 db per octave on either side of the transmission band. These compensation circuits have a gain of 1 while having an appropriate phase margin.
Has a slope closer to the corner frequency with a greater amplitude than the slope at frequencies near the critical frequency f _c of.

The compensation circuit of block 2 of FIGS. 5, 6 and 7 and 8 implements these principles. It can be seen that the loop compensation follows the guidelines from the open loop gain curve. In FIG. 4, the slope after the break frequency (500 Hz) is 18 db / octave. The network of section 2a has a zero at 80 Hz and 92H
It has a pole at z and damping constants of 0.56 and 1.1, respectively. The high-frequency circuit of section 2b has a zero at 3.1 KHz and a pole at 7.3 KHz, with damping constants of 0.49 and 1, respectively. The circuit and the low-pass filter of section 2c have the zero point of 160 Hz and 3 at 1.6 KHz and 3.4 KHz.
It has poles at 20 Hz, 800 Hz and 34 KHz.

The present invention has been described above according to the embodiments.
It will be apparent to those skilled in the art that many changes and modifications are possible within the scope of the invention.

[Brief description of drawings]

FIG. 1 (a) is shown as a cross-sectional view taken from line 1-1 in FIG. 1 (b) with a headphone according to the present invention worn on an ear.
(B) is a plan view of the headphone as seen from the ear.

FIG. 2 is a block diagram showing a logical configuration of a servo system according to the present invention.

FIG. 3 is a graph showing noise reduction measurements achieved by the present invention compared to logical noise reduction.

FIG. 4 is a graph showing the open loop gain and closed loop gain of a servo system on a common frequency scale.

FIG. 5 is a block diagram showing a logical configuration of a preferred embodiment of the present invention.

FIG. 6 is a block diagram showing a logical configuration of a preferred embodiment of the present invention.

FIG. 7 is a circuit diagram of an electrical circuit implementing the block diagrams of FIGS. 5 and 6;

FIG. 8 is a circuit diagram of an electrical circuit implementing the block diagrams of FIGS. 5 and 6;

FIG. 9 is a block diagram showing a preferred logical configuration of a compressor.

10 shows compression characteristics of the compressor shown in FIG.

11A and 11B are gain and phase characteristics of an ordinary operational amplifier having a first-order characteristic.

12A and 12B show characteristics obtained by modifying the characteristics shown in FIG. 11 according to the present invention.

FIG. 13 shows a bandpass response modified according to the present invention.

[Explanation of symbols]

 11 Microphone 12 Cavity 13 Housing 14 Diaphragm 15 Cushion 16 Outer ear 17 Driver 30 Signal combiner 31 Compressor 31A Compensation circuit 32 Power amplifier 35 Microphone preamplifier

Claims (9)

[Claims] 1. A driver means having a vibrating diaphragm for converting an input electric signal into an acoustic output signal; a housing means having a cushion support bracket for supporting the driver means; and a cushion support bracket. And a central opening defining a cavity for accommodating the diaphragm is formed to seal between the cushion support bracket and the outer ear and inhibit air flow between the cavity and an external area of the headphone device. Cushioning means for significantly attenuating spectral components in the intermediate frequency range, and electroacoustic conversion means for converting acoustic pressure signals in the cavity into corresponding converted electrical signals away from the driver means. A cavity in which the acoustic transducer is close to the diaphragm and remote from the cushion support. A headphone device in which the electroacoustic transducing means responds to pressure in the cavity near the ear sufficiently close to the end of the. 2. The headphone device according to claim 1, wherein a distance between the electroacoustic converting means and the diaphragm is smaller than a diameter of the diaphragm. 3. The headphone device of claim 2, wherein the distance between the electroacoustic transducing means and the end of the cavity remote from the cushion support bracket is less than the maximum span across the diaphragm. 4. A head horn device according to claim 1, wherein the axial cross-sectional area of said cavity is smaller than the axial cross-sectional area of the cushion means surrounding and defining said cavity. 5. A driver means having a vibrating diaphragm for converting an input electric signal into an acoustic output signal; a housing means having a cushion support bracket for supporting the driver means; and a cushion support bracket. And a central opening defining a cavity for accommodating the diaphragm is formed to seal between the cushion support bracket and the outer ear and inhibit air flow between the cavity and an external area of the headphone device. Cushion means for remarkably attenuating the spectrum component in the intermediate frequency range, and electroacoustic conversion means for converting the acoustic signal in the cavity into a corresponding electric signal, the electroacoustic conversion means being close to the diaphragm At the same time, sufficiently close to the end of the cavity away from the cushion support means, the electroacoustic conversion means In response to the pressure in the cavity close to the power amplification means for supplying an amplified signal to the driver means, and performing negative feedback of the converted signal to the power amplification means. And a means for canceling an acoustic signal entering the cavity from the outside of the headphone device by generating the signal component of the above in the cavity. 6. The negative feedback means comprises combining means for algebraically combining the negative feedback converted signal with an input signal characteristic of a desired acoustic signal to be reproduced in the cavity, further comprising: An input terminal coupled to the combining means for receiving the input electrical signal, and a combined signal supplied by the combining means for coupling to the power amplifying means to provide a cavity with an acoustic signal having a component representative of the input signal. 6. The headphone device according to claim 5, further comprising: providing means. 7. A means for coupling the combined signal to the power amplification means comprises compensation circuit means for achieving an open loop gain having a maximum gain over a predetermined low audio frequency and intermediate audio frequency range. The headphone device according to claim 6, which comprises: 8. The frequency range is about 40 to 2000H.
The headphone device according to claim 7, wherein z is z. 9. The negative feedback means is in a loop characterized by a loop gain, the loop reducing the loop gain in response to the input electrical signal and the converted electrical signal, and audible artifacts and A head horn device according to claim 5 comprising compressor means for preventing overload and audible distortion without producing oscillations.