WO2001033813A1

WO2001033813A1 - Communication terminal

Info

Publication number: WO2001033813A1
Application number: PCT/JP2000/007665
Authority: WO
Inventors: Taichi Majima
Original assignee: Kabushiki Kaisha Kenwood
Priority date: 1999-11-01
Filing date: 2000-10-31
Publication date: 2001-05-10
Also published as: JP3769433B2; JP2001136239A

Abstract

A communication terminal free from various problems attributed to the characteristics of the speech of the party, circuits, and the party's terminal and to the environment where the communication terminal is present and having a good speech quality. The communication terminal has two tone quality change modes: a noise mode for correcting the speech masking when the user of the communication terminal is suffering noise and a speech mode used when the speech of the party is difficult to hear in normal communication. Data stored in a memory (25) is read out and used to control a digital equalizer (15), a noise suppressor (16), a volume (17), an echo canceller (18), sidetone adding circuit (19) by means of an arithmetic unit (24) to set the communication terminal in the noise or tone quality mode.

Description

Specification

Communication terminal

Technical field

The present invention relates to a communication terminal such as a mobile phone, and more particularly to a sound quality adjustment function.

Background art

In the past, the adjustment of the sound quality of mobile phones was mostly one type. Also, even if a function that allows the user to select the sound quality is installed, adjustment of the frequency characteristics and adjustment of the noise suppression amount can only be made by selecting them from the menu. A technique for changing the sound quality in a communication terminal is disclosed in Japanese Utility Model Laid-Open Publication No. 7-442236, which changes the gain of a band-pass filter that divides a voice band into a plurality of parts, and changes the reception quality. To change the frequency characteristics of Also, for cordless telephones, Japanese Patent Laid-Open Publication No. Heisei 5-9-1168 discloses a technology that changes the sound quality of the received voice received and enhances the high frequency range so that the voice can be easily heard. ing.

Conventionally, when a mobile phone is used under noise, it is difficult for the user to hear the sound due to the surrounding noise, and the user has increased the volume or closed one ear.

To solve this problem, the system automatically detects the sound pressure of the surrounding noise and adjusts the sound pressure of the terminal, or creates an extra-large volume mode.

However, with the spread of mobile phones, the environment in which the user or the other party talks is diverse, and only the frequency characteristics can be changed or the noise suppression function can be uniformly changed depending on the noise state around the speaker or the communication line route. Alone is satisfactory Poor call quality. Also, during the call, you may feel dissatisfied. However, the prior art was not sufficient to solve these problems.

In the case of conventional menu selection, the user does not know under what circumstances and what characteristics should be changed, and other technologies are also widely used in current call terminals and used in various environments. Was unable to cope with the change in sound quality caused by the sound.

In addition, the following problems cannot be solved by adjusting only the volume of the conventional technology.

Under noise, the voice of the other party and the voice of the other party 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 is not only uncomfortable, but may be heard again due to sound quality deterioration due to sound cracking. In addition, the other party could hear a loud voice from the handset, resulting in suppression of vocalization and a vicious cycle such as making the call more difficult.

Just making the volume oversized will make your voice inaudible and easier to talk to, but will not completely solve the situation of screaming. Also, if the sound quality of the other party is difficult to hear, simply increasing the sound quality will mask the necessary components of the voice in the noise component.

Also, since the voice (sending) sent to the other party was not adjusted, the other party was in a situation where it was difficult to hear.

The present invention has been made in view of the above points, and has various characteristics generated from the voice quality of the other party, the characteristics of the line and the other party's terminal, and the environment where the user is located. It is an object of the present invention to provide a communication terminal that can eliminate a problem, obtain good call quality, and provide a comfortable call.

Disclosure of the invention

In order to achieve the above object, the communication terminal of the present invention provides an operation mode (noise mode) for compensating voice masking under noise and an operation mode (sound quality) for increasing the clarity or naturalness of a call sound. Mode) and one of these noise mode and sound quality mode can be set selectively.

The noise mode and the sound quality mode are each defined by a combination of predetermined characteristic values of a plurality of signal characteristics, and the plurality of signal characteristics include, for example, a frequency characteristic, a noise suppression degree, a volume, It is selected from the degree of echo cancellation and the amount of sidetone addition. For example, when the noise mode is set, the low frequency band and the high frequency band are emphasized, and when the sound quality mode is set, the high band is further emphasized in the narrow band of the middle frequency. Alternatively, the frequency characteristics are adjusted so that the entire frequency band is flattened. The communication terminal of the present invention includes a digital processing unit including a plurality of signal processing units for adjusting each of a plurality of types of signal characteristics, and includes a predetermined characteristic for each of the plurality of signal characteristics. The digital signal processing unit is controlled so as to selectively provide a characteristic pattern corresponding to the noise mode or the sound quality mode, which is a combination of values.

As the plurality of signal processing units, a digital equalizer, a noise suppressing circuit, a volume adjusting circuit, an echo removing circuit, a side tone adding circuit, and the like are used.

According to an aspect of the present invention, the signal processing unit is provided with a predetermined signal processing unit. Either the operation mode in which the first characteristic pattern is operated (clear mode) or the operation mode in which the processing unit is operated to provide the second predetermined characteristic pattern (natural mode). In the clear mode, it provides a frequency characteristic that emphasizes the high frequencies in the narrow band of the middle frequency, enhances the noise suppression function, further reduces sidetones, and increases the natural sound. In this mode, a signal characteristic value is selected that flattens the frequency response, reduces the noise suppression effect, and gives a moderate sidetone.

Further, the signal processing unit is operated to provide a third characteristic pattern that provides an intermediate characteristic pattern between the first characteristic pattern and the second characteristic pattern. The operation mode can be set.

According to another aspect of the present invention, each characteristic value of the plurality of signal characteristics is selected so as to optimize the quality of call sound under noise, and a low frequency band is set for reception. In addition to adjusting the frequency characteristics to emphasize the frequency, add sidetones, enhance the noise suppression effect for transmission, and adjust the signal characteristic value to suppress the others while leaving the center frequency band of the voice. You have to choose.

In the communication terminal of the present invention, an adjustment data defining a specific characteristic value is stored in a memory in advance for each of a plurality of signal characteristics, and is read out from the memory every time the operation mode is set. I have.

The setting of each operation mode is performed by key operation or switch operation, and the operation mode that is set is displayed on the display. BRIEF DESCRIPTION OF THE FIGURES

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 for explaining the operation at the end of a call in the first embodiment of the present invention.

FIG. 4 is a flowchart for explaining a second embodiment of the present invention.

FIG. 5 is a flowchart for explaining the operation at the end of a call in the second embodiment of the present invention.

FIG. 6 is a flowchart for explaining a third embodiment of the present invention.

Embodiment of the Invention

Hereinafter, embodiments of a communication terminal according to the present invention will be described in detail with reference to the drawings.

The first embodiment uses digital signal processing, and when the user is dissatisfied with the received voice, noise suppression, frequency characteristics, volume, sidetone, and echo removal are collectively performed by a simple button operation. The purpose of this setting is to make the reception satisfactory for the user. In addition, by setting these processes collectively in the digital processing unit, setting is facilitated.

Currently, telephone calls have various sound qualities. The reasons are as follows: (1) Various operators and call systems are mixed due to the expansion of services, Each has a unique sound quality.

(2) Even if it is specified for one system, the sound quality differs depending on various terminal manufacturers.

(3) Call quality depends on the user's age, gender, preferences, and other factors. And the like.

In resolving these issues, significant improvements can be made by introducing two concepts that are considered important when talking about sound quality. When describing sound quality, emphasizing the content of the conversation and raising the clarity, which indicates its clarity and recognizable degree, and the naturalness, which reproduces fine sounds closer to nature, will lead to improved sound quality. However, these two attributes have conflicting properties.

To increase clarity

(1) The frequency characteristics should be as close as possible to the communication band, and low and high frequency noise should be cut.

(2) Enhance the vicinity of 2 kHz in the speech band to increase the recognition level.

(3) Strengthen the noise suppression function to suppress components other than conversations sent from the other party.

(4) Sidetones should be reduced or completely eliminated, and only the other party's voice will be output from the speaker.

When increasing naturalness

(1) The frequency characteristics are almost flat at any frequency.

(2) Minimize or eliminate noise suppression to make any sound more audible to details.

(3) Set the sidetone to an appropriate volume, and bring it closer to a normal conversation. In order to satisfy the above two conditions, each setting is required, and simply changing the frequency characteristics cannot be satisfied. Therefore, in order to satisfy these conditions, multiple signal processing units such as a digital equalizer, a noise suppression circuit, a volume, an echo canceling circuit, and a sidetone adding circuit are adjusted at a time, so It can reproduce various characteristics of. In addition, by preparing an intermediate setting between these settings and these two settings that are usually used for mobile phones, etc., and allowing the user to select these three types of settings during a call, the call quality is greatly improved. Can be improved. If these settings increase the amount of echo generated by the terminal, set the echo cancellation circuit.

The first embodiment is realized by the mobile phone shown in FIG. In other words, the digital equalizer 15, the noise reducer 16, the poly unit 17, the echo canceller 18, and the sidetone adding circuit 19 are controlled by the arithmetic processing unit 24, and the adjustment data for control is Remember in memory 25.

The adjustment data in the memory 25 include the adjustment parameters of the equalizer 15, the suppression adjustment data of the noise sub-lesser 16, the volume data, the echo canceller data, and the adjustment data of the sidetone addition circuit 19. Prepare etc.

As shown in the flowchart of FIG. 2, 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 S21). S22). Then, each time the key is pressed, the sound quality setting changes. That is, every time the key is pressed, “clear”, “natural”, and “normal” (in any order) sound quality adjustment data are sequentially switched and read out from the memory 25 (step S23). The DSP 13 is controlled by the processing unit 24 with the 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 to use something that is easy for the user to understand, such as "clear mode, natural mode, normal mode".

At the end of the call, always control DSP 13 with normal adjustment data. That is, as shown in FIG. 3, when a termination request is issued (step S31), the normal adjustment data is read from the memory 25 (step S32), and the arithmetic processing is performed at that time. The DSP 13 is controlled by the device 24 (step S33). After that, the end call process in step S34 is performed.

In the above example, the side tone addition unit 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 above-described first embodiment, the following effects can be obtained.

(1) During a call, if the user is dissatisfied with the call quality due to the sound quality characteristics of the other party's telephone or the voice quality of the other party, the user can easily select the sound quality as desired by simple operation.

(2) Various sound qualities can be changed by providing a setting that raises the intelligibility and a setting that raises the naturalness, and a setting that is usually used in between.

(3) For a call that emphasizes the conversation, the conversation becomes smoother by using the setting with higher clarity.

(4) If you want to listen to various sounds faithfully using the information dial of a music program, etc., you can listen to the details of the sound by using a setting with increased naturalness. <Second embodiment>

In the second embodiment, digital signal processing is used, and when the user is under noise, the noise suppression, frequency characteristics, volume, sidetone, and echo removal are set collectively by simple button operation. The purpose of the present invention is to make transmission / reception talks optimal for use in a noise environment. In addition, setting is facilitated by providing those processes collectively in the digital signal processing unit.

When talking in a noisy environment, it is assumed that the following phenomena will occur due to psychological acoustic effects.

Listening (the sound you are listening to)

(Reception 1) The other party's received sound cannot be heard due to the masking effect of the noise.

(Reception 2) Your voice is masked by the masking effect of the noise, and the utterance volume increases.

(Receiver 3) Speak loudly while listening repeatedly, cover one ear or hold your hand over the microphone.

Transmission (sound heard by the other party)

(Send 1) The other party's voice is too loud.

(Send 2) The surrounding noise of the other party is large and my ear hurts.

(Send 3) Since the other party's voice is loud, respond with a quiet voice with psychological effects. When the other party listens again, they respond loudly.

To solve the above-mentioned reception 1, the reception volume is required several dB higher than the average value of the ambient noise. In addition, it is necessary to enhance the high-frequency component of the sound with high noiseness and the low-frequency part that is susceptible to masking to increase the clarity. Receiver 2 is solved by adding a sidetone that returns its voice from the handset.

Resolving the above two points will also solve the problem.

The problem of sending 1 is eliminated by solving the receiver. In transmission 2, the noise suppression of the transmission is set stronger, and the transmission frequency characteristic is narrowed to 500 Hz to 2 kHz, which is the main component of the voice, and a band-pass filter that suppresses other bands is applied. It is solved by doing. Reducing the volume slightly is also effective. Transmission 3 will disappear if these problems are solved.

However, it is conceivable that a new echo is generated by increasing the receiving volume. This can be solved by using the echo cancellation function strongly.

The second embodiment is realized by the mobile phone shown in FIG. That is, the digital equalizer 15, the noise reducer 16, the poly-um 17, the echo canceller 18, and the sidetone adding circuit 19 are controlled by the arithmetic processing unit 24, and the adjustment data for control is stored in the memory 25. Remember.

The adjustment data in the memory 25 include the adjustment parameters of the equalizer 15, the suppression adjustment data of the noise reducer 16, the adjustment data of the noise reducer 16, the adjustment of the echo canceller data, and the adjustment of the sidetone addition circuit 19. Prepare data etc. By controlling the transmission and reception with these data as described above, the optimum operation state under noise is obtained.

When the user wants to select the optimal operating condition under noise (hereinafter referred to as noise mode) during the call in step S41 of the flowchart shown in FIG. 4, the key input section is shown in step S42. 2 Operate the 7 selection key to select the noise mode. Then the noise from memory 25 Adjustment data is read out (step S43), the DSP 13 is controlled by the arithmetic processing unit 24 with this data (step S49), and a noise mode is set. At the same time, the display on the LCD display 29 is switched (step S45) to indicate that it is in the noise mode.

To release the noise mode, the user operates the release key of the key input section 27 (step S46). Then, 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 S48), whereby 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 is released. At the end of the call, the DSP 13 is controlled at all times for normal use. That is, when an end request is issued in step S51 of FIG. 5, the normal adjustment data is read from the memory 25 (step S52), and the DSP 13 is controlled with this data (step S53). ). After that, the call termination processing of step S54 is performed.

Note that, in this example as well, the sidetone addition unit and the volume adjustment were all performed inside the DSP 13, but other circuits may be used. Also, the equalizer can be realized by another circuit.

According to the second embodiment, the following effects can be obtained.

(1) By selecting when the surroundings of the station platform or crowded intersection are noisy, the conversation of yourself and the other party will be smoother, and you will not obstruct one ear or loudly.

(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 phone or a PHS. This is a case where the present invention is applied to a mobile communication terminal including (a personal handy phone system), but the present invention can also be applied to a radio device for business use and a hobby. Further, the present invention is applicable to fixed telephones, public telephones, and the like.

The third embodiment is based on the characteristics of the other party's line and telephone that the user feels during a call, and the sound quality dissatisfaction that makes it difficult to make a call based on the voice quality of the other party. The purpose of this is to eliminate each volume dissatisfaction that the other party's voice cannot be heard. Dissatisfaction with the sound quality can be caused by the fact that the other party's voice is muffled or that the content of the call is difficult to understand due to the other party's call line or environment, or vice versa. Occurs when is not playing. For the former (for voice), it is only necessary to use a setting in which the voice component is emphasized and other noise components are weakened. Can be realized. The latter (in the case of music) can be resolved by providing a setting that allows all sounds to be heard naturally, and can be realized by making the frequency characteristics broadband flat and weakening noise suppression or not at all. You. This setting is called "sound quality mode". 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 noisy, etc., and the reception volume is oversized, the frequency band is emphasized in the high / low range, and side sounds are added. This can be solved by making it possible to hear one's own voice. In addition, the transmitted sound can be corrected by correcting the voice sent to the other party by narrowing the frequency band and increasing the noise suppression. Call is made smoothly. This setting is called "noise mode". For the above two modes, “sound quality mode” and “noise mode”, it is easier to use different transition operations in consideration of the usage conditions. It is better to be able to change the setting of the sound quality mode while listening to the incoming call sound, so that the sound quality can be switched by operating one key. In addition, it is necessary to set the noise mode easily because the volume changes drastically, and to prevent malfunction. Considering this, noise mode can be set by pressing and holding the volume up key, which is usually 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 To do. Also, when using earphones or hands-free, sudden volume changes may cause an accident.

If the ambient noise is loud, it is not possible to hear fine sound quality, and it is unlikely that sound quality dissatisfaction will occur. Let me go back.

In addition, when a call is started, it starts in a normal state (a state other than either mode), and even if settings are changed during a call, it returns to a normal state when the call ends. This is based on the premise that each mode is used when you feel dissatisfied during a call.If the previous settings remain during the next call, you need to restore them. This is to omit.

Also, if you return to the call after FAX or PC communication during a call, keep the previous state.

Note that specific phone numbers often require specific sound quality adjustments. available. That is, for music information programs and telephone numbers that only call out specific persons, it is more convenient to store the sound quality adjustment data for each number. The state at the end of the call may be stored so that the next time the call is made, the call can be made from that state. FIG. 1 is a circuit diagram showing a mobile phone implementing the third embodiment. The radio wave received by the antenna 11 is supplied to the RF circuit 12 for frequency selection and the like. The output signal is supplied to a modem (Digital Signal Processor) 13 and a modem, and is converted into a digital signal for voice code conversion. Next, a plurality of signal processing units, namely, a digital equalizer 15, a noise suppression circuit (noise sub-lesser) 16, a volume 17, an echo removal circuit (echo canceller) 18, and a sidetone addition circuit 1 Go through 9 sequentially. 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 voice input from the microphone 22 is converted into a digital signal by the AZD converter 23 and transmitted as a radio wave by the antenna 11 in the reverse procedure.

In this mobile phone, a digital equalizer 15, a noise reducer 16, a volume 17, an echo canceller 18, and a side tone adding circuit 19 are controlled by an arithmetic processing unit 24, and adjustment data for control is stored in a memory 25. To memorize it.

The adjustment data in the memory 25 includes the adjustment parameters of the equalizer 15, the suppression adjustment data of the noise sub-resorer 16, the volume data, the echo canceller data, and the adjustment data of the sidetone adding circuit 19. Prepare etc. In each of the above-described modes of sound quality and noise, the control data of each circuit stored in the memory 25 is integrated to realize each mode by a combination thereof. Table 1 below shows the combination patterns. When each mode is set with the received data R (0) and the transmitted data T (0) in the normal state as defaults, the receiving side sets different data R (1), R (2 R (3 ), And the transmitting side changes the set value only to the noise mode to data T (1) .Two other sound quality modes correspond to the above-mentioned sound quality mode 1 and sound quality mode 2. In Fig. 1, 26 is a ROM that stores programs, 27 is a key input section that performs various inputs and settings, and 29 is a telephone number and settings. The LCD display 29 displays the status, and is controlled by the arithmetic processing unit 24 via the LCD driver 28.

FIG. 6 is a flowchart when the operation of the first embodiment is performed on the mobile phone. To explain this flowchart, during a call in step S1 (at the start of the call, the reception mode and the transmission data are in the standard mode with R (0) and T (0) in Table 1). It is determined whether the earphone is being used or the hands-free operation is being performed (step S2). If so, the change to the noise mode is prohibited, and the process proceeds to step S3. Then, the key is pressed in step S3, and a change to the sound quality mode is required. Then, the process proceeds to step S4 to switch the reception data to R (l), R (2) or R (0) every time the key is pressed, and to switch the mode 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 or sound quality mode 2 or standard mode) is displayed on the LCD display 29. Then, if 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 call is terminated in step S6, and the mode is set to the sound quality mode 1 In the case of or 2, the received data and transmitted data are returned to R (0) and T (0), and the mode returns to the standard mode. On the other hand, if the change to the sound quality mode is not required in step S3, the call is continued in the standard mode, and if the end key is pressed in step S5, the end process of step S6 is performed. . During the call in step S1 or the call in step S5 due to the end key not being pressed, it is determined in step S2 whether the earphone is being used or the hands-free operation is in progress. Proceed to S7 to enable change to noise mode. If the volume increase key is pressed and held 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 display that the sound mode is set to the noise mode. After displaying on the display unit 29, the reception time is set to R (3), the transmission time is set to T (l), and the noise mode is set (step S9). Even if it is set to mode 1 or 2, this sound quality mode ends here temporarily.) Thereafter, if the volume down key is pressed in step S10, or if the headphones are used, or if the handset is in the hands-free state, the process proceeds to step S11 where the transmission data is set to Τ (0), and Return to the previous mode, and return to the mode before the noise mode, and go to step S3. If the volume down key is not depressed in step S10, or the phone is not in use, or is not in the hands-free operation state, the process proceeds to step S12, and if the end key is depressed (step S12) ) Perform the call termination process in step S6 described above. If the volume up key is not pressed and held in step S7 and the operation does not proceed to the noise mode, the operation also proceeds to step S3. Step S3 and subsequent steps are the same as described above.

Instead of the above key operations, use the rotary switch to turn it, turn it to the sound quality mode, press and hold it for the noise mode, or use the single source switch to make the sound quality mode up and down. Press and hold to switch to noise mode. In addition, a rotary switch or seesaw switch can be used to release the noise mode.

According to the third embodiment described above, the dissatisfaction with the telephone, environment, and voice quality of the other party who seems dissatisfied during the call with the mobile phone, and the reason for being in a place with a lot of ambient noise, etc. The volume dissatisfaction that occurs can be eliminated. In addition, the telephone can be set smoothly and easily to a state where it can be canceled.

Industrial applicability

As described above in detail, according to the communication terminal of the present invention, good communication quality can be achieved by eliminating various problems caused by the voice quality of the other party, the characteristics of the line and the other party terminal, and the environment in which the user is placed. And the call becomes comfortable.

Claims

Scope of claim. In a communication terminal capable of adjusting the sound quality of a call sound, a first operation mode for compensating voice masking under noise and a second operation mode for increasing the clarity or naturalness of the call sound. A communication terminal having an operation mode, wherein one of the first and second operation modes can be selectively set. The communication terminal according to claim 1, wherein the first and second operation modes are defined by a combination of predetermined characteristic values of a plurality of signal characteristics, respectively. The communication terminal according to claim 2, wherein the plurality of signal characteristics are selected from a frequency characteristic, a degree of noise suppression, a volume, a degree of echo cancellation, and an amount of sidetone added. 4. The communication terminal according to claim 3, wherein at least one or both of the low frequency band and the high frequency band are emphasized when the first operation mode is set, and the second operation mode is set. A communication terminal whose frequency characteristics are adjusted so that, when setting the frequency, the narrow band of the middle frequency band is further emphasized in the narrow band, or the entire frequency band is flattened. A communication terminal having a digital signal processing unit for adjusting sound quality of a call sound, wherein the digital signal processing unit includes a plurality of signal processing units for adjusting each of a plurality of types of signal characteristics. A combination of predetermined characteristic values of the plurality of signal characteristics to selectively provide a characteristic pattern for compensating voice masking under noise and a characteristic pattern for increasing the clarity or naturalness of a call sound. A communication terminal for controlling the signal processing unit so as to be able to do so. The communication terminal according to claim 5, wherein the plurality of signal processing units are a digital equalizer, a noise suppression circuit, a volume adjustment circuit, an echo removal circuit, and a side tone addition circuit. The communication terminal according to claim 5, wherein the signal processing unit is operated to provide a predetermined first characteristic pattern, and the processing unit is set to a second operation mode. §1 end that allows any one of the second operation modes to operate to provide the characteristic pattern of the first embodiment. 8. The communication terminal according to claim 7, wherein in the first operation mode, a low-high frequency band is cut, a frequency characteristic for emphasizing a middle frequency band is given, and a noise suppressing function is enhanced. In the second operation mode, communication that flattens the frequency characteristics, reduces the noise suppression effect, and selects a signal characteristic value that gives an appropriate sidetone is reduced in the second operation mode. Terminal. The communication terminal according to claim 7, further comprising: a signal processing unit configured to provide a third characteristic pattern that provides an intermediate characteristic pattern between the first characteristic pattern and the second characteristic pattern. A communication terminal that allows you to set the third operation mode to operate. 0. The communication terminal according to claim 5, wherein each characteristic value of the plurality of signal characteristics is selected so as to optimize the quality of a call sound under noise.

1. The communication terminal according to claim 10,

For reception, adjust the frequency characteristics to emphasize the low-frequency and high-frequency, add sidetones, enhance the noise suppression effect for transmission, and increase the frequency band of the voice component. A communication terminal that selects the characteristic value to leave.

2. The communication terminal according to claim 5,

A communication terminal in which sound quality adjustment data defining a specific characteristic value for each of the plurality of signal characteristics is stored in a memory in advance, and is read from the memory each time an operation mode is set.

3. The communication terminal according to claim 12,

A communication terminal that stores the sound quality adjustment data in the memory in association with the other party's telephone number and reads the data from the memory at the next call.

4. The communication terminal according to claim 1 or 5,

Communication terminal where each operation mode is set by key operation or switch operation.

5. The communication terminal according to claim 1 or 5,

Communication terminal that displays which operation mode is set on the display unit.

6. The communication terminal according to claim 14,

A communication terminal in which each operation mode can be switched to another operation mode by an independent key or switch operation.

7. The communication terminal according to claim 14,

A communication terminal that shifts to a predetermined operation mode when a key or switch is pressed for a long time.

8. The communication terminal according to claim 14,

A communication terminal that releases a specific operation mode with a key or switch and returns the volume to its original state.

9. The communication terminal according to claim 14,

A communication terminal in which a particular mode of operation is performed with a single key or switch operation.

0. In the communication terminal according to claim 1 or 5,

A communication terminal that, when a specific mode is selected, prohibits other modes from being selected and does not accept them.

1. The communication terminal according to claim 1, 5, or 10,

The first operation mode is not accepted when using the earphones or the hands-free function, and the first operation mode is used when the use of the earphones or the hands-free function occurs during the first operation mode. Communication terminal that is designed to cancel.

2. The communication terminal according to claim 1 or 5,

A communication terminal that returns to the normal setting (default) at the end of a call, and starts with the default at the next call.

3. The communication terminal according to claim 22,

A communication terminal which, after shifting to the second operation mode and set, does not return to the default state when shifting to the communication mode again after shifting to the data communication mode.