JP3226121B2 - Intercom equipment - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a two-way simultaneous communication apparatus suitable for use in, for example, a video conference system or a so-called hands-free telephone.

[0002]

2. Description of the Related Art Two-way simultaneous communication devices such as a video conference system and a hands-free telephone are very convenient because they are hands-free and have both hands free so that they can perform writing and other work even during a call.

[0003] However, since the receiver is not attached to the ear as in the headset type, the volume radiated from the speaker must be appropriately increased. Similarly,
Since the transmitter is not located in the mouth of the speaker, the microphone must also have high sensitivity. For this reason, the level of the received signal from the speaker tends to be higher than the level of the voice of the sender, and the other party is mixed and listened to, so that it is very difficult to hear. Further, if the loop gain in the talking state is large, howling may occur.

In order to alleviate this problem, a variable gain device called a voice switch is often used in a hands-free telephone device. When the transmission level falls below a preset threshold (that is, when this side is not speaking), the apparatus increases the gain of the receiving side, raises the level of the receiving voice from the other side, and conversely, When the talk level becomes larger than the threshold, the gain on the receiving side is reduced.

[0005]

However, the call using the voice switch is not a complete two-way simultaneous call, but only when the other party is talking and the other party does not talk. It is valid. In addition, it is difficult to set a threshold value for switching the gain, and the received voice tends to be a voice with truncated words, and the conversation is unclear.

SUMMARY OF THE INVENTION In view of the above, it is an object of the present invention to provide a communication device capable of performing perfect two-way simultaneous communication with good conversation sound quality.

[0007]

In order to solve the above-mentioned problems, a communication device according to the present invention is provided with a speaker 31 for reproducing a received voice, and Placed between the sender and
A first microphone 11 for picking up a voice from a speaker , and a microphone 31 disposed between the speaker 31 and the speaker;
It is, picks up the received voice from the speaker 31, and,
Direction with low sensitivity to the voice input direction from the talker
A second microphone 21 having characteristics, and an adaptive filter circuit 24 for adaptively controlling an output audio signal of the second microphone 21 so as to approximate a received voice from the speaker 31 mixed into the first microphone 11.
And the output signal of the first microphone 11 and the output signal of the adaptive filter circuit 24,
From the speaker 31 in the output signal of the microphone 11
A synthesizing circuit 15 for reducing a received voice signal component, transmitting means 17 for transmitting an output signal of the synthesizing circuit 15 as a transmitted voice signal, the speaker 31, the first and second microphones 11, 21 Case 4 to which is mounted
0 is provided.

[0008]

According to the present invention having the above-described structure, the voice of the sender and the reproduced voice from the speaker 31 are collected by the first microphone. Is reduced or eliminated from the transmitted voice signal transmitted to the other party.

[0009]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS In the present invention, the concept of adaptive noise reduction processing is used, so before describing one embodiment of the present invention, the adaptive noise reduction processing will be described.

FIG. 5 is a block diagram of the basic configuration of the adaptive noise reduction processing system, wherein 1 is a main input terminal, 2 is a reference input terminal, and a main input signal input through the main input terminal 1 is a delay circuit. The signal is supplied to the synthesizing circuit 4 via the line 3. The delay circuit 3 detects a time delay in the adaptive filter circuit 5 when there is no time delay between the main input signal input to the main input terminal 1 and the reference input signal input to the reference input terminal 2. The delay circuit 3 does not need to be provided.

The signal input through the reference input terminal 2 is supplied to the synthesizing circuit 4 via the adaptive filter circuit 5, and is subtracted from the signal from the delay circuit 3. And
The output of the synthesizing circuit 4 is fed back to the adaptive filter circuit 5 and is output to an output terminal 6.

In this noise reduction device, a signal obtained by adding a desired signal s and uncorrelated noise n0 to the main input terminal 1 is input. On the other hand, the noise n1 is input to the reference input terminal 2. This reference input noise n1
Are uncorrelated with the desired signal, but are correlated with the noise n0.

The adaptive filter circuit 5 has a reference input noise n1.
To output a signal approximating the noise n0, that is, a signal y having the same phase and the same amplitude as the noise n0. As the output signal of the adaptive filter circuit 5, a signal -y having the same phase and the same amplitude as the noise n0 can be obtained.
The synthesizing circuit 4 performs a process of subtracting the output signal of the adaptive filter circuit 5 from the output signal of the delay circuit 3 (adding when the output signal is a signal -y having a phase opposite to that of the noise n0).

The adaptive algorithm in the adaptive filter circuit 5 works to minimize the subtraction output (residual output) e output from the synthesis circuit 4. That is, now s, n
Assuming that 0, n1, y are statistically stationary and the average value is 0, the residual output e is e = s + n0-y. The expected value of squaring this is that s is n0,
Also, since it is uncorrelated with y, Becomes Assuming that the adaptive filter circuit 5 converges,
The adaptive filter circuit 5 is adjusted so that E [e ² ] is minimized. At this time, since E [s ^2] is not ^{affected, Emin [e 2] = E} [s 2] + Emin [(n0 -
y) ² ]. That is, by minimizing E [e ² ], E [(n 0 −y) ² ] is minimized, and the output y of the adaptive filter circuit 5 becomes an estimated amount of the noise n 0. And
The expected value of the output from the combining circuit 4 is only the desired signal s. That is, adjusting the adaptive filter circuit 5 to minimize the total output power is equivalent to the fact that the subtraction output e becomes the least square estimation value of the desired audio signal s.

In general, the residual output e is a signal s with some noise remaining, but since the output noise is given by n0-y, E [(n0-y) ² ] is minimized. This is equivalent to maximizing the signal-to-noise ratio of the output.

The adaptive filter circuit 5 can be realized either by an analog signal or a digital signal processing circuit. FIG. 6 shows an example in which the adaptive filter circuit 5 is realized using a digital filter. This example uses a so-called L as an adaptive algorithm.
This is an example when the MS (least mean square) method is used.

As shown in FIG. 6, in this example, an FIR filter type adaptive linear combiner 300 is used. this is,
A plurality of delay elements DL1, DL2,... DLm (m is a positive integer) each having a delay time Z ^-1 of a unit sampling time, an input noise n1, and each delay element DL1, DL
, And DLm, and a weighting circuit MX0, MX1, MX2,..., MXm for multiplying the output signals of the weighting coefficients, and an addition circuit 310 for adding the outputs of the weighting circuits MX0 to MXm. The output of the adder 310 is y (or -y).

The weighting coefficients supplied to the weighting circuits MX0 to MXm are formed on the basis of the residual signal e from the synthesizing circuit 4 by an LMS operation circuit 320 composed of, for example, a microcomputer. The algorithm executed by the LMS operation circuit 320 is as follows.

Now, the input vector X _k at time k is
As shown in FIG. 6, X _k = [x _0k x _1k x _2k ... X _mk ] ^T , the output is y _k , and the weighting factor is w _jk (j = 0, 1, 2,... M). Then, the relationship between input and output is as shown in the following Equation 1.

[0020]

(Equation 1) If the weight vector W _k at time k is _defined as W _k = [w _{0 k} w _{1 k} w _{2 k} ... W _mk ] ^T , the input / output relationship is given by y _k = X _k ^T · W _k . Here, assuming that a desired response is d _k , the residual e _k is expressed as follows. In the _{e k = d k -y k =} d k -X k T · W k LMS method, the update of the weight vector, by _{W k + 1 = W k +} 2μ · e k · X k ...... (a) becomes formula Go sequentially. Here, μ is a gain factor (step gain) that determines the speed and stability of adaptation.

Next, FIG. 1 shows a block diagram of one embodiment of the communication apparatus according to the present invention using the above-described adaptive noise reduction processing. In this example, a digital filter is used as the adaptive filter circuit, and a subtraction circuit is used as the synthesis circuit 4. FIG. 2 shows the appearance of the communication device of this example.

In FIG. 1, reference numeral 11 denotes a main input microphone for mainly collecting a desired voice, that is, a voice of a transmitter in this example. Reference numeral 31 denotes a speaker that emits a voice received from the other party. Reference numeral 21 denotes a reference input microphone for collecting an unnecessary sound to be removed as noise, that is, a received sound emitted from the speaker 31 in this example.

As shown in FIG. 2, the microphones 11, 21 and the speaker 31 are mounted on a housing 40 of the communication device. The housing 40 is a flat and thin type so-called low-profile type. FIG. 3 is a diagram showing a use state of the communication device. The communication device is placed on the table 41 for the sender 10 as shown. That is, as shown in FIG. 3, the main input microphone 11 is attached to the front end position of the low-profile housing 40 closest to the transmitter 10 in this use state. The speaker 31
At a position away from the microphone 11 of the housing 40,
Moreover, it is mounted at a position farther than the transmitter 10. The reference input microphone 21 is a speaker 31 in this example.
And a main input microphone 11.

As shown in FIG. 3, the communication device is placed on the table 41 so that the voice of the sender 10 as the desired sound enters the housing 40 of the communication device from the direction indicated by the arrow AR. Mounted and used. The main input microphone 11 is connected to the voice from the speaker 10 and the speaker 31.
An omnidirectional (omnidirectional) microphone as shown in FIG.

On the other hand, the reference input microphone 21 does not pick up the sound from the speaker 10 but picks up only the sound from the speaker 31, so as shown in FIG. It is composed of a unidirectional microphone that has no sensitivity in the direction of arrival of the sender's voice indicated by the arrow AR.

The communication apparatus of this example performs two-way simultaneous transmission and reception of a conversation via a telephone line L. That is,
A telephone line L is connected to the telephone circuit 18, and a signal sent from the other party via the telephone line L is supplied to the receiving circuit 32 via the telephone circuit 18, and the received voice signal is reproduced. This is supplied to the speaker 31 via the output amplifier 33, and the received voice is emitted from the speaker 31. And the voice of the sender 10 is the microphone 11
The transmission voice signal is formed as described below, and the transmission voice signal is transmitted from the transmission circuit 17 to the telephone line L via the telephone circuit 18 and transmitted to the other party. You.

That is, the voice of the conversation of the transmitter 10 and the voice from the speaker 31 are collected by the main input microphone 11 and converted into electric signals. The output audio signal of the microphone 11 is supplied to an A / D converter 13 via an amplifier 12, converted into a digital signal, and supplied to a subtraction circuit 15 via a delay circuit 14. The delay circuit 14 is for compensating for a time delay such as a propagation time in the adaptive filter circuit 24 and a time delay required for an operation for adaptive processing.

The sound from the speaker 31 picked up by the reference input microphone 21 is converted into an electric signal, supplied to an A / D converter 23 via an amplifier 22, converted into a digital signal, and adapted. Filter circuit 2
4 is supplied.

In this embodiment, the adaptive filter circuit 24
Is composed of an FIR filter type adaptive linear combiner 300 as shown in FIG. 6 and an arithmetic circuit for adaptively controlling the linear combiner 300, in this example, an LMS arithmetic circuit 320. The digital signal from 23 is supplied to the subtraction circuit 15 via the linear combiner 300. The adaptive filter circuit 24 can be configured by a DSP (digital signal processor) including a microprocessor.

The output signal of the subtraction circuit 15 is fed back to the arithmetic circuit 320 of the adaptive filter circuit 24,
The signal is returned to the analog signal by the / A converter 16 and supplied to the transmission circuit 17.

Basically, in the adaptive filter circuit 24,
Control is performed so that the reference input audio signal approximates the audio from the speaker 31 included in the main input audio signal. The voice of the sender 10 in the voice collected by the main input microphone 11 and the voice from the speaker 31 are uncorrelated (the transmitted voice and the received voice are generally almost uncorrelated. In the subtraction circuit 15, the audio signal from the adaptive filter circuit 24 is subtracted from the audio signal from the main input microphone 11 to cancel the audio from the speaker 31 in the main input audio. Is done. Accordingly, only the voice signal of the voice of the sender 10 that is the desired voice is obtained from the subtraction circuit 15.

That is, in the basic configuration of this embodiment, an output audio signal of the main input microphone 11 is supplied as a main input, and an output audio signal of the reference input microphone 21 is supplied as an unnecessary signal as a reference input. It has a configuration of an adaptive noise reduction system. In this example, the voice signal from the speaker 31 is selectively removed, and as a result, only the voice signal of the transmitter, which is the desired voice, is supplied to the transmission circuit 17 with a required quality. You. In other words, super directivity has been realized as an adaptive system.

In the above example, a unidirectional microphone is used as the reference input microphone 21 so that the desired voice of the sender is not picked up as much as possible. Microphone 21
It is difficult to make the directional characteristics of the desired voice signal not to completely pick up the desired voice signal, so that the desired voice signal of the sender is mixed into the reference input signal at a certain level. .

This state departs from the precondition for adaptive processing that both input signals of the desired voice and the reference input voice are uncorrelated. In particular, when the level of the unnecessary signal in the reference input signal is extremely low, it is the same as the case where a very small residual noise is input to the reference input, and the characteristics of the adaptive filter circuit 24 greatly deviate from the originally desirable one. Would. Since the system works to minimize the output power, the desired transmitted signal will also suffer some distortion. That is, when the received voice is at a low level that can be regarded as not present, in the above-described ordinary adaptive processing, the voice of the sender, which is the desired voice, may be reduced.

The example shown in FIG. 7 is an example of a communication device in which the desired voice itself can be reduced and distortion can be prevented.
That is, in this example, the level detection circuit 34 for detecting the level of the reception voice signal from the reception circuit 32 is provided. Then, the output of the level detection circuit 34 is supplied to the LMS operation circuit 320 of the adaptive filter circuit 24. If the level of the received voice signal is low enough that it can be considered that no received voice exists, the adaptive filter The adaptive processing in the circuit 24 is restricted. The rest of the configuration is exactly the same as in the example of FIG.

The limitation of the adaptive processing is as follows.
This is done by restricting the updating of the weighting factor in. The following three methods are used to limit the updating of the weighting coefficient.
There are seeds. (1) Set the step gain μ = 0. (2) The step gain μ is made smaller than a normal value, for example, 1/10. (3) Set the step gain μ = 0 and set the filter coefficient value to a default value prepared in advance.

In the methods (1) and (2), the updating of the coefficient in the adaptive filter circuit 24 is stopped or suppressed, so that the coefficient at the time when the received signal is at an appropriate level is again reduced to the level of the received signal. Are held until the threshold value in the level detection circuit 34 becomes an appropriate value larger than the threshold value. According to this method, a significant change in the operation of the adaptive filter circuit 24 due to the transmission signal and the residual noise can be suppressed.

In the method (3), for example, the speakers 31 to 31
A transfer function between the microphones 11 and 21 is measured in advance, and its coefficient is loaded to suppress a received signal. The transfer coefficient can be measured at the time of product shipment (that is, adjustment at the factory), but it is better to perform the adaptive operation using white noise immediately before using the communication device of this example.

By the above-described limitation of the adaptive operation, it is possible to prevent the desired audio signal from being distorted when the level of the reference input is lowered.
Normal adaptive processing is performed in the adaptive filter circuit 24, so that unnecessary signals in the main input audio signal can be appropriately reduced.

Incidentally, the reference input microphone 21
In general, a unidirectional microphone unit is used, but the unidirectional microphone is expensive, its characteristics are easily changed when installed, and its characteristics need to be adjusted. On the other hand, an omnidirectional microphone unit is inexpensive and small, and is easy to handle and adjust. By using a plurality of the non-directional microphone units, desired directional characteristics can be realized. An example in which two unidirectional microphone units are used to realize a unidirectional microphone will be described with reference to FIGS. 8 and 9.

As shown in FIG. 8, in this example, the non-directional microphone units 51 and 52 are arranged at a distance d. Then, as shown in FIG.
From the outputs of the microphone units 51 and 52, an output having a unidirectional characteristic is formed in the unidirectional forming circuit 50. That is, in the circuit 50, the output audio signal of the one microphone unit 51 is supplied to the subtraction circuit 53 via an amplifier (not shown). An output audio signal of the other microphone unit 52 is supplied to a subtraction circuit 53 via an amplifier and a filter circuit 54, which are also not shown. The filter circuit 54 includes a resistor 55 and a capacitor 56 in this example. When the resistance of the resistor 55 is R1 and the capacitance of the capacitor 56 is C1, the resistance R1 and the capacitance C1 are selected so that C1.R1 = d / c (where c is the speed of sound). ing.

In this example, an output audio signal is derived from the output of the subtraction circuit 53 to an output terminal 58 via a frequency characteristic correction circuit 57 such as an integrator for flattening the frequency characteristic. As will be described later, the frequency characteristic correction circuit 57 is provided as needed, and need not be provided.

The operation of the microphone of this example will be described. As shown in FIG. 8, the sound source is located at an angle of θ with respect to the arrangement direction of the two microphone units 51, 52, and the two microphone units 51, 5
2, each unit 51, 52
^Are P0 and P1, the output P1 is P1 = ^{P0.epsilon.-j.omega. (D / c) cos .theta} . Here, ω is an angular frequency.

Since the output of the microphone unit 52 is supplied to the subtraction circuit 53 through the filter circuit 54, the output signal Pa of the subtraction circuit 53 is as shown in the following equation (2).

[0045]

(Equation 2) In Equation 2, A represents a filter function of the filter circuit 54, and ω · d / c << 1.

If the following equation (3) is satisfied in equation (2), the output Pa indicates unidirectionality.

[0047]

(Equation 3) That is, to satisfy the following equation (3), the number 2, Pa = P0 · A · jω (d / c) (1 + cosθ) , and becomes a unidirectional angularly theta.

By the way, in the case of the above example, the filter function A of the filter circuit 54 is represented by A = 1 / (1 + jωC1 ・ R1), and is configured so that C1 ・ R1 = d / c. A = 1 / (1 + jωd / c), and from Equation 3, it is clear that the microphone of the embodiment of FIG. 8 has unidirectionality. However, the frequency characteristic of this microphone becomes a characteristic that rises to the right (response increases as the frequency increases). In this example, the frequency characteristic correction circuit 5
Reference numeral 7 is provided to correct the upward-sloping characteristic flat.

In the example of FIG. 9, the filter circuit 5
4. Subtraction circuit 53, and frequency characteristic correction circuit 57
Can also be realized by a digital filter or a processing program (software).

For example, as shown in FIG. 10, the filter circuit 54 can be constituted by a digital filter including an adder 61, a delay circuit 62, and a feedback amplifier 63 having a transfer function A.

In the example shown in FIG.
The two microphones mounted on the microphone 0 are both constituted by omnidirectional units 11U and 21U, and the main input sound uses the signal from the unit 11U as it is. It is obtained from a circuit 50A having the same configuration as the circuit 50. That is, units 11U and 2U
The 1U output signal is supplied to the unidirectional forming circuit 50A via the amplifiers 12 and 22, respectively.
From A, as shown in the lower part of FIG. 12, similarly to the above-described reference input microphone 21, the voice arrival direction AR from the transmitter 10 has no sensitivity, and the direction of the speaker 31 is the main axis. An output audio signal SA having a unidirectional characteristic as a direction is obtained. Then, the audio signal SA is supplied to the A / D converter 23 as a reference input audio signal.

Also in this example, the distortion of the voice of the sender when the received signal level is lowered is prevented. The monitoring of the reception signal level is not the method of the above-described example of monitoring the level of the input signal of the speaker 31, but the level of the output signal of the unidirectional reference input microphone whose main axis is directed to the speaker 31. It can also be done by However, in actual use, a part of the transmission signal is slightly mixed into the reference input due to reflection of a room or a nearby object.

On the other hand, it is extremely difficult to accurately detect each of the level of the reception signal and the level of the transmission signal, which leads to an increase in size and cost of the apparatus. Therefore, in this example, a decrease in the level of the received signal is estimated by monitoring the level ratio between the received signal and the transmitted signal.

In the case of this example, the transmitted voice for monitoring the level ratio is also obtained as an output voice having a unidirectional characteristic having the maximum sensitivity in the direction AR (the main axis direction is AR). For this reason, two microphone units 11U
And 21U are supplied to a unidirectional forming circuit 50B via amplifiers 12 and 22, respectively.
12B, as shown in the upper part of FIG. 12, contrary to the above-described reference input microphone 21, the output sound signal SB having the maximum sensitivity in the sound arrival direction AR from the transmitter 10 having the maximum sensitivity. Is obtained.

Then, the received voice output SA from the unidirectional forming circuit 50A and the transmitted voice output SB from the unidirectional forming circuit 50B are supplied to the level ratio detection circuit 35, and the outputs SA and SA are output. The output level of the SB is detected, and the output level ratio (received signal level / transmitted signal level) is detected. Then, the detection circuit 35 supplies the control signal to the arithmetic circuit 320 of the adaptive filter circuit 24, and performs a normal adaptive process when the level ratio of the reception signal level / transmission signal level is equal to or more than the set value. hand,
The received signal in the transmitted signal is suppressed. When the level ratio is lower than the set value, the adaptive operation is limited in the same manner as described above.

The microphone 11 (or 11)
The arrangement of U), 21 (or 21U) and the speaker 31 is not limited to the case where they are arranged linearly in the direction AR as in the above example.

FIG. 13 is a diagram showing an example of the arrangement of the microphones 11 and 21 and the speaker 31. That is, FIG.
3A is the arrangement of the example described above, and the microphones 11 and
This is an example in which the speaker 21 and the speaker 31 are linearly arranged. FIG.
In the example of 3B, the microphones 11 and 21 are connected to the talker 10.
It is an example of arranging at the same distance from the camera. FIG. 13C is an example in which the microphone 21 is arranged close to the speaker 31. FIG. 13D is an example in which the microphone 21 is arranged on the side of the speaker 31. In short, the microphone 11
May be arranged between the speaker 31 and the transmitter 10, and the microphone 21 may not be behind the speaker 31.

By the way, the configuration of the speaker 31 as an audio output device used in this type of communication device uses a thin speaker unit and uses a small speaker unit whose main axis is directed almost vertically. It is realized by turning the main axis in the direction where the listener's ear will come.

FIG. 14 shows an example of the configuration in the above case.
FIG. 3 shows a cross-sectional view of a state where the speaker unit 71 is attached to a thin enclosure (housing) 72. The main axis of the speaker unit 71 is oriented substantially vertically. FIG. 15 shows an example of the configuration in the above case, and is a cross-sectional view showing a state where the main axis of the speaker unit 73 is attached to the enclosure 74 so as to be substantially directed toward the ear of the listener.

However, in the above method, the listener's ear receives the sound at an angle of, for example, 60 ° with respect to the main axis of the speaker unit 71. For this reason, sound waves radiated from each part of the speaker unit 71 interfere with each other, and the sound pressure in the high frequency range is remarkably reduced along with the angle thereof. As a result, the directivity becomes sharp in the high frequency range, and the position of the listener is increased. The clarity and the like will change depending on this. FIG. 16 shows a small unit 71 having a diameter of 5 cm in FIG.
Install as shown, measure about 50cm in familiar position
Frequency characteristics measured by installing a microphone for
Expressed based on the frequency characteristics of θ = 0 °
is there.

Also, in the above method, the size (diameter) of the speaker unit that can be used is limited, and the use of a small unit significantly reduces the volume, especially the reproduction ability in the low frequency range. Conversely, if an attempt is made to ensure a certain level of reproduction capability, the size of the speaker unit becomes large, and it is inevitably impossible to provide a low-profile device suitable for the external shape of this type of communication device. Furthermore, the reflected wave from the desk surface causes interference with the direct wave, causing a decrease in sound pressure in the middle and high frequency range, and the clarity and the like also deteriorate.

FIG. 17 shows a small unit 73 having a diameter of 5 cm.
As shown in FIG. 15, the frequency characteristic is measured by placing the measuring microphone 76 at a position about 50 cm away from the main axis of the unit 73 on a desk 75 at an angle of about 60 °. In this case, the frequency characteristics in a state where there is no symbol are represented based on the frequency characteristics.

As is apparent from FIG.
There is a valley, especially in the midrange.
The peaks and valleys are generated by interference between the direct wave radiated from the unit 73 and directly reaching the measurement microphone 76 and the reflected wave arriving at the desk 75, and basically, the peak of the direct wave and the reflected wave is generated. A valley occurs at a frequency at which the difference in path length is an odd multiple of a half wavelength of the sound wave. The frequency may fluctuate depending on the size of the desk and the directional characteristics of the speaker unit. It is generally said that if this valley occurs in the midrange, the sound will not be clear.

In order to avoid the above-mentioned drawbacks, in the example described below, the size of the sound output radiation port is reduced, and the position of the radiation port is minimized on the surface of the desk on which the communication device is placed. Get closer.

In order to reduce the size of the sound output radiation port, for example, as shown in FIG.
A baffle 78 having a hole 77 having a smaller diameter than the diameter of the speaker unit 71 may be attached to the front surface of the speaker unit 71. By doing so, the size of the sound source becomes smaller than the wavelength, and broad directional characteristics close to non-directionality can be obtained. FIG.
Is a baffle 7 having a hole 77 having a diameter of 2 cm smaller than the diameter of the unit 71 on the front surface of the small unit 71.
8 is a directional frequency characteristic expressed on the basis of the characteristic of θ = 0 ° in a state where 8 is attached. It can be seen from the comparison with FIG. 16 that the directional characteristics are almost omnidirectional. Normal,
When this device is used on a desk, the listener's ear canal becomes θ
= 30 ° in many cases, and the broad directional characteristic around this means that even when the listener's position changes, there is little change in timbre and the listener can listen.

In order to bring the sound emission port close to the desk,
As shown in the figure, a low-profile enclosure
What is necessary is just to attach a speaker unit. In the sound emission port,
Further, an acoustic waveguide (horn) may be attached, and the radiation port of the horn may be installed near the desk surface. When the sound emission port approaches the desk surface, the path difference between the direct sound reaching the listener's ear and the reflected sound from the desk surface becomes smaller, and the valley on the frequency characteristic due to interference is relegated to the higher frequency range, and clarity is improved. Can be heard without deterioration.

FIG. 20 shows an example of the appearance of a communication device according to the present invention, in which an audio output device taking the above into consideration is mounted on its housing. In the communication device of this example, the speaker unit 81 is attached to a low-profile housing 82. In the housing 82,
A hole 83 having a diameter smaller than the diameter of the speaker unit 81 is formed, and the center of the hole 83 is aligned with the center of the speaker unit 1. 2, microphones 11 and 21 are attached to the front end of the housing 82 on the listener (sender) side. The sound wave emitted from the mouth of the sender becomes a direct wave to the microphone 11 and a reflected wave from the surface of the desk. , And good frequency characteristics with few peaks and valleys can be obtained.

With the above configuration, the sound wave emitted from the speaker unit 81 and radiated from the hole 83 exhibits almost non-directionality as described above, and particularly, the listener (speaker) moves. Even if the position of the ear canal changes, the sound pressure frequency characteristic does not change much.

Further, even when the apparatus is placed on a desk, the position of the hole 83 is low because the case 82 is of a low profile, and the path difference between the direct wave and the reflected wave to the ear canal is small. Mountains and valleys are difficult to create, or even if they are, they are relegated to treble.

In FIG. 20, the speaker unit 81
Is mounted with a slight inclination, but this
This is because the smoother frequency characteristics can be obtained by bringing the end on the listener (sender) side 2 closer to the desk surface.

[0071]

As described above, according to the present invention, a noise reduction system is constituted by using a transmission signal as a main input and voice from a speaker for reproducing a reception signal as a reference input signal. The output signal can be only a transmission signal from the transmitter. Therefore, a good two-way simultaneous call state can be maintained.

According to the present invention, since the reception signal in the transmission signal is adaptively suppressed, the optimum state can be maintained independently even when the environment around the communication device changes. In addition, when the level of the received signal is low, the adaptive operation is restricted, so that the transmitted signal itself can be prevented from being distorted.

In a video conference system, when a speaker for reproducing a received voice is installed on the TV side,
In this invention, the speaker is physically separated from the microphone on the terminal side of the communication device, so that the size of the speaker can be reduced. Since the directions match, a natural call can be made.

[Brief description of the drawings]

FIG. 1 is a block diagram of an embodiment of a communication device according to the present invention.

FIG. 2 is a diagram showing an example of an appearance of a communication device according to the present invention.

FIG. 3 is a diagram showing a use mode of one embodiment of the communication device according to the present invention.

FIG. 4 is a diagram for explaining the directional characteristics of a microphone used in the communication device according to the present invention.

FIG. 5 is a block diagram illustrating an outline of an adaptive noise reduction system.

FIG. 6 is a diagram illustrating an example of an adaptive filter circuit of the adaptive noise reduction system.

FIG. 7 is a block diagram of another example of the communication device according to the present invention.

FIG. 8 is a diagram for explaining an example of obtaining a predetermined directional characteristic using a plurality of microphone units.

FIG. 9 is a diagram for explaining an example in which a predetermined directional characteristic is obtained using a plurality of microphone units.

FIG. 10 is a diagram showing another example of a part of the configuration of FIG. 9;

FIG. 11 is a block diagram of another example of the communication device according to the present invention.

FIG. 12 is a characteristic diagram used for explaining the operation of the example of FIG. 11;

FIG. 13 is a diagram showing an arrangement example of a main input microphone, a reference input microphone, and a speaker used in the communication device according to the present invention.

FIG. 14 is a diagram illustrating an example of a sound output device used for a communication device.

FIG. 15 is a diagram showing another example of the sound output device used for the communication device.

16 is a frequency characteristic diagram of the sound output device of the example of FIG.

17 is a frequency characteristic diagram of the sound output device of the example of FIG.

FIG. 18 is a diagram for explaining an audio output device used as an example of the communication device according to the present invention.

FIG. 19 is a frequency characteristic diagram of the sound output device of the example of FIG. 18;

FIG. 20 is a diagram showing another example of the appearance of the communication device according to the present invention.

[Explanation of symbols]

Reference Signs List 10 Sender 11 First microphone 15 Subtraction circuit 17 Transmission circuit 18 Telephone circuit 21 Second microphone 24 Adaptive filter circuit 31 Speaker 32 Receiving circuit 34 Level detection circuit 35 Level ratio detection circuit 40, 72, 82 Housing 50 , 50A, 50B Unidirectional forming circuits 71, 73, 81 Speaker units 77, 83 Holes smaller in diameter than speaker units

────────────────────────────────────────────────── ─── Continuation of the front page (72) Inventor Kaoru Gyokudori Sony Corporation 6-35 Kita-Shinagawa, Shinagawa-ku, Tokyo (56) References JP-A-60-102052 (JP, A) 4-108246 (JP, A) JP-A-63-299435 (JP, A) JP-A-63-37748 (JP, A) JP-A-1-198154 (JP, A) JP-A-4-59696 (JP, A) U) Actually open sho 59-111385 (JP, U) (58) Field surveyed (Int. Cl. ⁷ , DB name) H04M 1/00-1/23 H04M 1/58-1/62

Claims (5)

(57) [Claims] 1. A speaker for reproducing a received voice, a first microphone disposed between the speaker and a speaker for collecting voice from the speaker , the speaker and the speaker. A speaker for receiving the voice from the speaker , and
A second directional characteristic having low sensitivity to the voice input direction
From the output audio signal of the second microphone,
An adaptive filter circuit that adaptively controls so as to form a signal approximating a received voice from the speaker mixed into the microphone, an output signal of the first microphone, and an output signal of the adaptive filter circuit. a synthesizing circuit for reducing reception voice signal component from the loudspeaker in the output signal of the first microphone, and transmitting means for transmitting the output signal of the combining circuit as transmission signal, and the speaker, the A communication device comprising: a housing on which the first and second microphones are mounted; 2. The communication device according to claim 1, wherein the level of the received voice supplied to the speaker is a predetermined level.
Detection means for detecting that the received voice level is equal to or less than the value.
When it is detected that the value falls below a predetermined value, the adaptive
Reduce the gain factor for updating coefficients in filter circuits
The communication device. 3. A speaker for reproducing a received voice, and a speaker disposed between the speaker and a transmitter.
The first is placed at a predetermined distance on the line
And the second omnidirectional microphone and the output of the first and second omnidirectional microphones
Et al., Low sensitivity to sound voice input direction from the talker
First unidirectional forming circuit for obtaining output of unidirectional characteristics
And the output of the first and second omnidirectional microphones
In this case, the direction of the voice input from the sender is set as the main axis direction.
A second unidirectional shaping circuit for obtaining an output of one directivity characteristic, and the speed from the output of the first unidirectional shaping circuit.
A signal similar to the received voice signal component from
An adaptive filter circuit for adaptively controlling the output signal of the first microphone;
And the output signal of the first micro-phone.
Audio signal component from the speaker in the output signal of the
And a transmission for transmitting an output signal of the synthesis circuit as a transmission signal
Means, the speaker, and the first and second omnidirectional microphones
Call device comprising a housing in which the Rohon is mounted. 4. The communication device according to claim 3, wherein an output of said first unidirectional forming circuit is connected to said second unidirectional forming circuit.
Level for detecting the level ratio with the output of the unidirectional shaping circuit
Ratio detecting means, wherein the level ratio detected by the level ratio detecting means is
When the value falls below a predetermined value, the coefficient of the adaptive filter circuit
A communication device in which the gain factor for updating is reduced . 5. A are grave to the housing, the diameter of the opening hole for speech sound of the speaker, the communication apparatus according to any one of claims 1 to 4 so as to be smaller than the wavelength of the received speech sound .