JP2001136239A - Communication terminal - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a communication terminal capable of obtaining excellent speech communication quality by eliminating various problems generated by the voice quality of the opposite side, the characteristics of the line and the opposite terminal, the environment where a self is placed, etc. SOLUTION: Two voice quality change modes are prepared. One is a noise mode that corrects voice masking when a speaker exists under noise and the other is a voice quality mode used in case the opposite voice is hard to listen to in a normal speech communication mode. It is possible to set the communication terminal to the noise mode or the voice quality mode by reading data stored in a memory 25 and in such a manner that an operation processor 24 controls a digital equalizer 15, a noise suppressor 16, a volume 17, an echo canceler 18 and a sidetone additional circuit 19 with the data.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a communication terminal such as a portable telephone, and more particularly to a sound quality adjusting function.

[0002]

2. Description of the Related Art Hitherto, the adjustment of the sound quality of a portable telephone has been mostly one type. Even when a function that allows the user to select the sound quality is provided, adjustment of the frequency characteristics and adjustment of the noise suppression amount can only be provided in the menu. As a technique for changing the sound quality in a communication terminal, there is a technique described in Japanese Utility Model Laid-Open Publication No. Hei 7-42236, which changes the gain of a band-pass filter that divides a voice band into a plurality of parts, and adjusts the frequency characteristics of a received voice. It changes. Also, in the case of a cordless telephone only, a technology for changing the sound quality of a received voice received and making the sound quality emphasized in a high frequency range to make the voice easier to hear is disclosed in Japanese Patent Application Laid-Open No. 5-911.
No. 68 ”. Conventionally, when a mobile phone is used in a noisy environment, it is difficult for the user to hear sound due to ambient noise, and the user has increased the volume or closed one ear. In order to solve this problem, the system automatically detects the sound pressure of the surrounding noise and adjusts the sound pressure of the terminal.

[0003]

However, with the widespread use of mobile phones, the environment in which a user or the other party talks is various, and only the frequency characteristics can be changed or changed depending on the noise state around the speaker or the communication line route. Satisfactory speech quality cannot be obtained with a uniform noise suppression function alone. Also,
During a call, you may feel dissatisfied with the call because of the situation where the other party's phone is placed or the voice quality of the other party dissatisfied with the sound quality of the reception and the noise of the surrounding environment where you talk The prior art was not sufficient to solve these problems. In addition, in the case of the conventional menu selection, the user does not know what kind of environment to change what characteristics, and other technologies also have the sound quality due to the spread of current call terminals and use in various environments. Was unable to keep up with the changes.

Further, the following problems cannot be solved by adjusting only the volume in the prior art. Under noise, the voice of the other party, and even the voice of oneself, cannot be heard due to the masking effect of the ambient noise, making it difficult to talk. In addition, the voice of the user becomes loud, and the other party may not only be uncomfortable, but also re-listen due to sound quality deterioration due to sound cracking. Further, since the other party can hear a loud voice from the handset, the utterance is suppressed, and a vicious cycle such as making the call more difficult has sometimes occurred. Just by making the volume oversized, you can't hear your voice,
It becomes easier to talk, but the situation of shouting cannot be completely solved. Further, if the sound quality of the other party is difficult to hear, simply increasing the sound quality masks the necessary components of the voice in the noise component. Also, since the voice (sending) side to be sent to the other party is not adjusted, the other party is in a situation where it is difficult to hear.

[0005] The present invention has been made in view of the above points,
To provide a communication terminal that can obtain good call quality by eliminating various problems that occur from the voice quality of the other party, the characteristics of the line and the other party's terminal, the environment where the user is located, etc., and make the call comfortable. With the goal.

[0006]

According to a first aspect of the present invention, there is provided a communication terminal capable of adjusting sound quality during a call, comprising: a noise mode for correcting voice masking when a caller is in a noise; It is characterized by having two sound quality change modes of a sound quality mode used when it is difficult to hear the other party's voice during a call. A second communication terminal according to the present invention is a communication terminal having a digital signal processing unit, wherein the digital signal processing unit includes a digital equalizer, a noise suppression circuit, a volume adjustment unit, an echo removal circuit, and a sidetone adding circuit. By controlling them by a processing unit during a call, the operation state is selectively set to an operation state in which the clarity of the received sound is increased or an operation state in which the naturalness is increased. A third communication terminal of the present invention is a communication terminal having a digital signal processing unit, wherein the digital signal processing unit has a digital equalizer, a noise suppression circuit, a volume adjustment unit, an echo removal circuit, and a sidetone addition circuit inside the digital signal processing unit. The operation state is set to the optimum operation state for the time of noise by controlling them by the arithmetic processing unit at the time of noise.

[0007]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Embodiments of a communication terminal according to the present invention will be described below in detail with reference to the drawings. <First Embodiment> In the first embodiment, it is difficult to make a telephone call due to the characteristics of the other party's line and telephone and the voice quality of the other party that the user feels during the telephone conversation, It is an object of the present invention to solve each volume dissatisfaction that the other party cannot hear the voice of the other party due to the noise state or the like. Dissatisfaction with the sound quality is caused by the fact that the other party's voice is muffled or that it is difficult to understand the contents of the call due to the other party's call line or environment, or vice versa, when the sound of a music service etc. is heard, the details of the sound are reproduced Occurs if not. In the former case, it is only necessary to use a setting in which the voice component is emphasized and other noise components and the like are weakened, and this can be realized by setting the frequency characteristic to a narrow band treble emphasis type to strongly apply noise suppression. The latter can be solved by providing a setting that allows all sounds to be heard naturally, and can be realized by making the frequency characteristics flat in a wide band so that noise suppression is weakened or not suppressed at all. This setting is called “sound quality mode”. Further, the former setting is called “sound quality mode 1”, and the latter setting is called “sound quality mode 2”. Volume dissatisfaction often occurs when the environment in which the user is placed is in a noisy state, etc. This can be solved by making it possible to hear the sound. Also, the voice transmitted to the other party can be corrected by making the frequency band narrower and the noise suppression stronger, so that the call can be smoothly performed. This setting is called “noise mode”.

The above two modes are easier to use if the transition operation is made different in consideration of the situation in which they are used. It is better that the sound quality mode can be changed while listening to the received sound. In this case, the sound quality is switched by operating one key. In addition, since the noise mode has a large change in sound volume, it is necessary to easily set the noise mode and to prevent malfunction. Considering this, the sound mode can be set by pressing and holding the volume up key normally used during a call, and release can be easily released by pressing the volume down key, and the volume returns to the volume before setting I do. Also, when using earphones or hands-free, there is a possibility that an accident may occur due to a sudden change in volume, so setting is not possible. If the ambient noise is loud, it is not possible to hear even the finest sound quality, and it is considered that sound quality dissatisfaction is unlikely to occur. I do. In addition, a call is started in a normal state (a state other than either mode) at the start of a call, and even if a setting is changed during a call, the call returns to a normal state at the end of the call. this is,
This is because it is assumed that each mode is used when you feel dissatisfied during a call, and it is necessary to return the previous setting when the next call is made, and to omit this operation. is there. Also, when returning to the call after performing a FAX or personal computer communication during the call, the previous state is maintained.

FIG. 1 is a circuit diagram showing a portable telephone which realizes the first embodiment. The radio wave received by the antenna 11 is supplied to the RF circuit 12, where frequency selection and the like are performed. This output signal is supplied to a modem (modulation / demodulation circuit) of a DSP (Digital Signal Processor) 13 and a modem, and is converted into a digital signal, voice code-converted, a digital equalizer 15, a noise suppression circuit (noise suppressor) 16,
Volume 17, echo removal circuit (echo canceller)
18, sequentially passing through the sidetone adding circuit 19. Then, the signal is converted into an analog signal by the D / A converter 20 and converted into a sound by the speaker 21. On the other hand, the sound collected by the microphone 22 is A
The signal is converted into a digital signal by the / D converter 23 and transmitted as a radio wave by the antenna 11 in a reverse procedure. In this mobile phone, a digital equalizer 15 and a noise suppressor 1
6, the volume 17, the echo canceller 18, and the side tone adding circuit 19 are controlled by the arithmetic processing unit 24, and adjustment data for control is stored in the memory 25. As adjustment data in the memory 25, adjustment parameters of the equalizer 15, suppression adjustment data of the noise suppressor 16,
Volume data, echo canceller data, adjustment data for the sidetone adding circuit 19, and the like are prepared. The sound quality described above,
In each mode of the noise, the control data of each circuit stored in the memory 25 is grouped to realize each mode by a combination thereof. Table 1 below shows the combination patterns. With the reception data R (0) and the transmission data T (0) in the normal state as defaults, when each mode is set, the reception side has different data R (1), R (2), R
Using (3), the transmitting side changes the set value to data T (1) only in the noise mode. The sound quality mode is assumed to be two in addition to the usual sound quality mode corresponding to the sound quality mode 1 and the sound quality mode 2 described above. In FIG. 1, 26 is a ROM for storing programs, 27 is a key input unit for performing various inputs and settings, and 29 is an LCD display for displaying telephone numbers and setting states. LCD display 29
Is controlled by the arithmetic processing unit 24 via the LCD driver 28.

[Table 1]

FIG. 2 is a flowchart when the operation of the first embodiment is performed in the portable telephone. This flowchart will be explained. During the call in step S1 (at the start of the call, the reception data and the transmission data are set to R in Table 1).
(0), T (0) is the standard mode), it is determined whether the earphone is being used or the hands-free operation is being performed (step S).
2) If so, change to the noise mode is prohibited, and the process proceeds to step S3. When the key is depressed in step S3 and a change to the sound quality mode is requested, the flow advances to step S4 to switch the reception data to R (1) or R (2) or R (0) every time the key is depressed. The mode is switched to the sound quality mode 1, the sound quality mode 2, or the standard mode. At the same time, the switched mode (sound quality mode 1, sound quality mode 2, or standard mode) is displayed on the LCD display 29. Then, when the end key is pressed during a call in the sound quality mode 1, the sound quality mode 2, or the standard mode (step S5), the end processing of step S6 is performed. The transmission data is returned to R (0), T (0) and returns to the standard mode. On the other hand, if the change to the sound quality mode is not requested in step S3, the call is continued in the standard mode and the process proceeds to step S5.
If the end key is depressed, the end call process in step S6 is performed.

During a call in step S1 or in a call due to the end key not being pressed in step S5,
In step S2, it is determined whether the earphone is being used or the hands-free operation is being performed. If not, the process proceeds to step S7 to enable the change to the noise mode. If the volume up key is pressed and held down in step S7, the flow advances to step S8 to store the reception data indicating the current sound quality mode state in the memory 25 and to set the sound mode to L.
After displaying on the CD display 29,
(3) The transmission data is set to T (1) and the noise mode is set (step S9) (thus, even if the sound quality mode is set to 1 or 2 in step S4, this sound quality mode is temporarily stopped here). Thereafter, when the volume down key is pressed in step S10, when the earphone is used, or when the hands-free state is set, step S1 is performed.
The process proceeds to 1 to return the transmission data to T (0), the previous data storing the reception data, return to the mode before the noise mode, and proceed to step S3. If the volume down key is not depressed in step S10, the earphone is not used, or it is not in the hands-free operation state, step S1
Proceed to step 2 and press the end key (step S12).
The call termination processing in step S6 described above is performed. Step S7
Also, when the volume up key is not pressed for a long time and does not proceed to the noise mode, the process also proceeds to step S3. Then, steps after step S3 are the same as described above.

Instead of the above key operation, a dial-type key is used, and when it is turned, it is in a sound quality mode, when it is long-pressed, it is in a noise mode. It can also be a mode.
Furthermore, a dial key or a seesaw switch can be used to cancel the noise mode.

According to the first embodiment,
It is possible to eliminate sound quality dissatisfaction caused by the telephone, environment, and voice quality of the other party who is dissatisfied during a call with the mobile phone, and volume dissatisfaction caused by being in a place with a lot of ambient noise. In addition, the telephone can be smoothly and easily set to a state in which it is canceled.

<Second Embodiment> A second embodiment uses digital processing, and when a user is dissatisfied with a received voice, noise suppression, frequency characteristics, sound volume, and volume can be controlled by a simple button operation. It is an object of the present invention to make the reception satisfactory for the user by setting the sound and echo removal collectively. In addition, the setting is facilitated by collectively providing these processes in the digital processing unit.

At present, telephone calls have various sound qualities. The reasons are as follows: (1) Various services and communication systems are mixed due to the expansion of services, and each has a unique sound quality. (2) Even if one system is specified, the sound quality differs depending on various terminal manufacturers. (3) The call quality is divided according to the user's age, gender, preference, and the like. And the like. In solving these, a considerable point can be improved by introducing two concepts that are considered important when describing sound quality. When describing the sound quality, emphasizing the content of the conversation and increasing the clarity indicating the clarity and recognizable degree and the naturalness that reproduces fine sounds closer to the natural world lead to the improvement of the sound quality. However, these two properties have conflicting properties. In the case of increasing clarity (1) The frequency characteristic should be as close as possible to the communication band to cut low- and high-frequency noise. (2) The vicinity of 2 kHz of the communication band is emphasized to increase the degree of recognition. (3) The noise suppression function is strengthened to suppress components other than conversation sent from the other party. (4) The side sound is reduced or completely eliminated so that only the other party's voice is output from the speaker. Increasing the degree of naturalness (1) Make the frequency characteristics nearly flat at any frequency. (2) Noise suppression is reduced or completely eliminated so that any sound can be heard in detail. (3) Set the sidetone to an appropriate volume and bring it closer to the state of ordinary conversation.

In order to satisfy the above two conditions, each setting is required, and it cannot be satisfied simply by changing the frequency characteristics. Therefore, by adjusting the digital equalizer, the noise suppression circuit, the volume, the echo removal circuit, and the sidetone adding circuit at a time to satisfy these conditions, various characteristics of the voice can be reproduced. In addition, by preparing an intermediate setting between these settings and these two settings usually used for a mobile phone, etc., and allowing the user to select these three types of settings during a call, the call quality can be greatly improved. Can be improved. If the amount of echo generated by the terminal increases due to these settings, the setting of the echo removing circuit is also performed.

The second embodiment is realized by the mobile phone shown in FIG. That is, the digital equalizer 15, the noise suppressor 16, the volume 17, the echo canceller 18, and the side tone adding circuit 19 are controlled by the arithmetic processing unit 24, and adjustment data for control is stored in the memory 25. As adjustment data in the memory 25, adjustment parameters of the equalizer 15, suppression adjustment data of the noise suppressor 16, volume data, echo canceller data, adjustment data of the side tone adding circuit 19, and the like are prepared. As shown in the flowchart of FIG. 3, the user operates a predetermined key of the key input unit 27 when changing the setting while feeling dissatisfied with the sound quality during the call in step S21 (step S22). Then, each time the key is pressed, the setting of the sound quality is switched. That is, every time a key is pressed, “clear”, “natural”, “normal” (in any order) is read from the memory 25.
Are sequentially switched and read out (step S23), the DSP 13 is controlled by the arithmetic processing unit 24 with this sound quality adjustment data (step S24), and the sound quality setting is changed. At the same time, the display on the LCD display 29 is switched (step S25), and the set sound quality mode is displayed. It is appropriate that the display is easy to understand for the user, such as “clear mode, natural mode, normal mode”. At the end of the call, the DSP 13 is always controlled with normal data. That is, as shown in FIG. 4, when a termination request is issued (step S31), the normal adjustment data is stored in the memory 25.
(Step S32), and the DSP 13 is controlled by the arithmetic processing unit 24 with the data (step S3).
3). Thereafter, the call termination processing of step S34 is performed.

In the above example, the side tone adding section and the volume adjustment are all performed inside the DSP 13, but they may be replaced by other circuits. Also, the equalizer can be realized by another circuit.

According to the second embodiment, the following effects can be obtained. (1) The user can select a desired sound quality by a simple operation when he or she feels dissatisfied with the sound quality of the other party's telephone or the voice quality of the other party during the call. (2) By providing a setting in which the clarity is increased and a setting in which the naturalness is increased, and an intermediate setting that is generally used, various sound qualities can be changed. (3) In the case of a call in which the conversation is particularly important, the conversation is smoothed by using the setting in which the clarity is increased. (4) When it is desired to listen to various sounds faithfully by an information dial of a music program or the like, it is possible to hear details of the sounds by using a setting with an increased naturalness.

<Third Embodiment> In a third embodiment, digital processing is used, and when the user is under noise,
By simply setting noise suppression, frequency characteristics, volume, sidetone, and echo removal by simple button operation,
It is an object of the present invention to make transmission / reception speech optimal for use in a noise environment. In addition, the setting is facilitated by collectively providing these processes in the digital processing unit.

When talking in a noisy environment, it is assumed that the following phenomena occur due to psychological acoustic effects. Receiving (sound heard by yourself) side (Reception 1) The receiving sound of the other party cannot be heard due to the masking effect due to noise. (Reception 2) The user's voice is masked by the masking effect of the noise, and the utterance volume increases. (Receiver 3) Make a loud voice while listening repeatedly, cover one ear or hold your hand over the microphone. Sender (sound heard by the other party) (Send 1) The other party's voice is too loud and broken. (Send 2) Ambient noise on the other side is large and my ear hurts. (Send 3) Since the other party's voice is loud, respond with a small voice with psychological effects. When the other party listens again, it responds loudly.

In order to solve the above-mentioned reception 1, a reception sound volume several dB higher than the average value of the ambient noise is required. It is also necessary to enhance the high-frequency component of the sound with strong noise and the low-frequency portion that is susceptible to masking to increase the clarity. Receiver 2 is solved by adding a sidetone that returns its voice from the handset. By resolving the above two points, the reception 3 is also resolved. The problem of sending 1 is eliminated by solving the receiver. Transmission 2 sets transmission noise suppression to a relatively high level and sets transmission frequency characteristics to 500 Hz to 2 kHz, which is a central component of voice.
The problem is solved by applying a band-pass filter that narrows down to z and suppresses other bands. It is also effective to slightly reduce the volume. Transmission 3 is eliminated if these problems are solved. However, it is conceivable that a new echo is generated by increasing the reception volume. This can be solved by strongly applying the echo canceling function.

The third embodiment is realized by the portable telephone shown in FIG. That is, the digital equalizer 1
5. The arithmetic processing unit 24 controls the noise suppressor 16, the volume 17, the echo canceller 18, and the sidetone adding circuit 19, and adjustment data for control is stored in the memory 25.
Remember. As adjustment data in the memory 25, adjustment parameters of the equalizer 15, suppression adjustment data of the noise suppressor 16, volume data, echo canceller data, adjustment data of the side tone adding circuit 19, and the like are prepared. By controlling transmission and reception based on these data as described above, an optimum operation state under noise is obtained. The user selects step S4 of the flowchart shown in FIG.
In order to select an optimal operation state under noise (hereinafter referred to as a noise mode) during the call 1, the user operates the selection key of the key input unit 27 to select the noise mode as shown in step S42. Then, the noise adjustment data is read out from the memory 25 (step S43), and the DSP 13 is controlled by the arithmetic processing unit 24 with this data (step S43).
49), the mode becomes the noise mode. At the same time, the LCD display 29
Is switched (step S45) to indicate that the mode is the noise mode. To release the noise mode, the user operates a release key of the key input unit 27 (step S46). Then, the normal adjustment data is read from the memory 25 (step S47), and the DSP 13 is controlled by the arithmetic processing unit 24 with this data (step S4).
8) The normal mode is set. At the same time, the display on the LCD display 29 is switched (step S49) to indicate that the noise mode has been released. At the end of the call, the DSP 13 is always controlled with normal data. That is, when an end request is issued in step S51 of FIG. 6, the normal adjustment data is read from the memory 25 (step S52), and the DSP 13 is controlled with this data (step S53). Thereafter, the call termination processing in step S54 is performed.

In this example, the sidetone adding section and the volume adjustment are all performed inside the DSP 13, but other circuits may be used. Also, the equalizer can be realized by another circuit.

According to the third embodiment, the following effects can be obtained. (1) By selecting when the surroundings such as the platform of the station or the crowded intersection are noisy, the conversation of oneself and the other party becomes smooth, and it is not necessary to cover one ear or make a loud voice. (2) There is also an effect of correcting a voice that is too low depending on the other party's line.

In each of the above embodiments, the present invention is applied to a mobile communication terminal including a mobile phone or a PHS (Personal Handy Phone System).
The present invention can also be used for business and hobby wireless devices. Further, the present invention can be used for fixed telephones, public telephones, and the like.

[0027]

As described in detail above, according to the communication terminal of the present invention, it is possible to eliminate various problems caused by the voice quality of the other party, the characteristics of the line and the other party terminal, and the environment where the user is located. Good call quality can be obtained, and the call becomes comfortable.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a mobile phone according to an embodiment of the present invention.

FIG. 2 is a flowchart for explaining the first embodiment of the present invention.

FIG. 3 is a flowchart illustrating a second embodiment of the present invention.

FIG. 4 is a flowchart for explaining an operation at the time of a call termination in a second embodiment of the present invention.

FIG. 5 is a flowchart for explaining a third embodiment of the present invention.

FIG. 6 is a flowchart for explaining an operation at the end of a call in the third embodiment of the present invention.

[Explanation of symbols]

 13 DSP (Digital Signal Processor) 15 Digital Equalizer 16 Noise Suppressor 17 Volume 18 Echo Canceller 19 Sidetone Addition Circuit 24 Arithmetic Processing Unit 25 Memory 27 Key Input Unit 29 LCD Display

──────────────────────────────────────────────────続 き Continued on the front page (51) Int.Cl. ⁷ Identification symbol FI Theme coat ゛ (Reference) H04M 1/60 H04B 7/26 Q 1/725 109F

Claims (17)

[Claims] 1. A communication terminal capable of adjusting sound quality during a call, comprising: a noise mode for correcting voice masking when the caller is in noise; and a sound quality mode for use when the other party's voice is difficult to hear during a normal call. A communication terminal having two sound quality change modes. 2. The communication terminal according to claim 1, wherein a shift operation between the noise mode and the sound quality mode is different. 3. The communication terminal according to claim 1, wherein the mode is shifted to the noise mode by a long press of a key or a switch. 4. The communication terminal according to claim 1, wherein the noise mode is canceled by a key or a switch, and the sound volume is returned to an original state. 5. The communication terminal according to claim 1, wherein the sound quality mode is executed by one key or switch operation. 6. The communication terminal according to claim 1, wherein the setting of the sound quality mode is prohibited when the noise mode is set, and the sound quality mode is not accepted during the noise mode. 7. The noise mode according to claim 1, wherein the noise mode is not accepted when the earphone is used or the handsfree mode is used, and when the earphone use or the handsfree mode is used during the noise mode, the noise mode is canceled. Communication terminal. 8. Normal setting at the end of every call (default)
The communication terminal according to claim 1, wherein the communication terminal is started by default at the next call. 9. After switching to the sound quality mode and setting, if there is no transition to a non-telephone or a call channel (switch back), the normal setting (default) is performed until a call termination operation is performed at the next call. The communication terminal according to claim 1, wherein the communication terminal is not returned. 10. A communication terminal having a digital signal processing unit, wherein the digital signal processing unit includes a digital equalizer, a noise suppression circuit, a volume adjustment unit, an echo removal circuit, and a side tone addition circuit, which are used during a call. A communication terminal which is selectively set to an operation state in which clarity of a received sound is increased or an operation state in which naturalness is increased by controlling by an arithmetic processing unit. 11. An operation state intermediate between two operation states of an operation state with increased clarity and an operation state with increased naturalness is added, and the operation state is selected from the three operation states. The communication terminal according to claim 10. 12. A communication terminal having a digital signal processing unit, wherein the digital signal processing unit has a digital equalizer, a noise suppression circuit, a volume adjustment unit, an echo removal circuit, and a side tone addition circuit in the digital signal processing unit. A communication terminal, wherein an operation state is set to an optimum operation state with respect to a time of noise by being controlled by an arithmetic processing unit. 13. The communication terminal according to claim 10, wherein the selection of the operation state or the setting of the optimum operation state is selected by a key operation. 14. The display device according to claim 10, wherein when the optimum operation state is set, which sound quality operation state is set or that the optimum mode is selected is displayed on a display. Communication terminal as described. 15. The data for adjusting the operation state are frequency characteristics, noise suppression, volume adjustment, echo removal, and sidetone addition, and these data are stored in a memory.
2. The method according to claim 1, wherein the data is read for each type of operation state.
13. The communication terminal according to 0 or 12. 16. Data to be adjusted to an optimum operation state are frequency characteristics, noise suppression, volume adjustment, echo removal, and addition of side sounds. These data are stored in a memory, and are set at the time of setting or mode selection. 13. The communication terminal according to claim 10, wherein the communication terminal is read from the communication terminal. 17. The apparatus according to claim 1, wherein the control for setting the optimum operation state is performed for transmission and reception.
Or the communication terminal according to 10 or 12.