WO2001033813A1 - Communication terminal - Google Patents
Communication terminal Download PDFInfo
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- WO2001033813A1 WO2001033813A1 PCT/JP2000/007665 JP0007665W WO0133813A1 WO 2001033813 A1 WO2001033813 A1 WO 2001033813A1 JP 0007665 W JP0007665 W JP 0007665W WO 0133813 A1 WO0133813 A1 WO 0133813A1
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- Prior art keywords
- communication terminal
- operation mode
- terminal according
- mode
- noise
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
Definitions
- the present invention relates to a communication terminal such as a mobile phone, and more particularly to a sound quality adjustment function.
- 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.
- 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.
- 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.
- 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.
- 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.
- 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.
- 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.
- 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.
- an operation mode noise mode
- an operation mode sound quality
- 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.
- the noise mode 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.
- 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.
- a digital equalizer 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.
- the signal processing unit is provided with a predetermined signal processing unit.
- 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).
- 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.
- a signal characteristic value is selected that flattens the frequency response, reduces the noise suppression effect, and gives a moderate sidetone.
- 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.
- 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.
- a low frequency band is set for reception.
- add sidetones enhance the noise suppression effect for transmission
- adjust the signal characteristic value to suppress the others while leaving the center frequency band of the voice. You have to choose.
- 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.
- 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.
- 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.
- setting is facilitated.
- the frequency characteristics should be as close as possible to the communication band, and low and high frequency noise should be cut.
- the first embodiment is realized by the mobile phone shown in FIG.
- 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.
- 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.
- 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".
- step S31 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.
- 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.
- 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.
- setting is facilitated by providing those processes collectively in the digital signal processing unit.
- Receiveiver 3 Speak loudly while listening repeatedly, cover one ear or hold your hand over the microphone.
- Receiver 2 is solved by adding a sidetone that returns its voice from the handset.
- the problem of sending 1 is eliminated by solving the receiver.
- 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.
- 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.
- 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.
- 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.
- noise mode 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.
- step S46 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.
- step S49 the display on the LCD display 29 is switched (step S49) to indicate that the noise mode is released.
- 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.
- 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.
- 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”
- 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.
- 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.
- noise mode This setting is called “noise mode”.
- sound quality mode and “noise mode”
- 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.
- a call 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.
- 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.
- modem Digital Signal Processor
- 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.
- the signal is converted into an analog signal by the D / A converter 20 and converted into a sound by the speaker 21.
- 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.
- 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.
- 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.
- 1, 26 is a ROM that stores programs
- 27 is a key input section that performs various inputs and settings
- 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.
- 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.
- 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.
- the switched mode (sound quality mode 1 or sound quality mode 2 or standard mode) is displayed on the LCD display 29.
- step S5 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.
- 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. .
- step S2 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.
- step S9 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 ⁇ (0), and Return to the previous mode, and return to the mode before the noise mode, and go to step S3.
- step S10 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.
- rotary switch 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.
- a rotary switch or seesaw switch can be used to release the noise mode.
- the volume dissatisfaction that occurs can be eliminated.
- the telephone can be set smoothly and easily to a state where it can be canceled.
- 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.
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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
明 細 書 Specification
通 信 端 末 Communication terminal
技術分野 Technical field
本発明は、 携帯電話機等の通信端末に係り、 特に音質調整機能に 関する。 The present invention relates to a communication terminal such as a mobile phone, and more particularly to a sound quality adjustment function.
背景技術 Background art
従来、 携帯電話機の音質の調整は 1種類であることがほとんどで あった。 また使用者が音質を選ぶことができる機能を搭載した場合 でも、 周波数特性の調整やノイズ抑圧量の調整をメニューに設け選 択できるだけである。 通信端末において音質を変更する技術として 実開平 7 — 4 2 2 3 6号公報に記載されたものがあり、 これは音声 帯域を複数に分割する帯域通過フィル夕の利得を各々変化させ、 受 話の周波数特性を変えるものである。 また、 コードレス電話機に限 れば受信された受話音声の音質を変更し高域を強調した音質にする ことで音声を聞き取りやすくする技術が特開平 5 — 9 1 1 6 8号公 報に記載されている。 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.
また、 従来、 携帯電話機を騒音下で使用する場合、 周囲の騒音に より音が聞きづらく使用者がボリュームを大きく したり、 片耳をふ さいだり していた。 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.
これを解決する手法としては周りの騒音の音圧を自動的に検知し 端末の音圧を調整するものや、 ボリ ューム特大モー ドを作って対応 していた。 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.
しかし、 携帯電話機の普及により使用者または通話相手が通話す る環境は様々であり、 話者周辺の騒音状態や通信回線の経路等によ り、 周波数特性のみの変更や一様なノィズ抑圧機能だけでは満足い
く通話品質が得られない状況にある。 また、 通話中、 不満に感じる こととして、 相手の電話機が置かれた状況や相手の声質により受話 の音質に不満を感じることと、 自分が通話する周囲の環境の騒音が 激しいなどの理由で通話が妨害されることがあり、 これらを解決す るには従来技術では十分ではなかった。 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.
また、 従来技術のメニュー選択の場合は、 どのような環境でどの 特性を変化させれば良いか使用者が分からず、 またその他の技術も 現在の通話端末の普及や様々な環境下での使用による音質の変化に 対応しきれない状態だつた。 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.
さらに、 従来技術のボリュームのみの調整では、 下記に挙げる問 題を解決できない。 In addition, the following problems cannot be solved by adjusting only the volume of the conventional technology.
騒音下では周囲ノイズによるマスキング効果により相手の声、 さ らには発声した自分の声が聞こえず通話は困難となる。 また、 使用 者の声は大きくなり、 相手側が不快なばかりでなく音割れによる音 質劣化のため聞き直されることもある。 さらに、 相手側は受話器か ら聞こえる声が大きいため発声の抑制が起こ り、 さらに通話を困難 にするなどの悪循環が起こっていることがあった。 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.
ボリュームを特大にしただけでは、自分の声は聞こえないままで、 話しやすくはなるが大声を張り上げる状態は完全に解決できない。 また相手の音質が聞き取りにく いものであると単に大きく しただけ では、 ノイズ成分に音声の必要な要素がマスクされてしまう。 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.
また、 相手側に送る音声 (送話) は調整されないので相手側も聞 きづらい状況にあった。 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.
本発明は上記の点に鑑みてなされたもので、 相手の声質、 回線や 相手端末の特性、 および自分の置かれた環境等から発生する各種の
問題を除去して良好な通話品質を得ることができ、 通話が快適とな る通信端末を提供することを目的とする。 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.
発明の開示 Disclosure of the invention
本発明の通信端末は、 上記目的を達成するために、 騒音下での音 声マスキングを補償する動作モー ド(騒音モー ド)および通話音の明 瞭度または自然度を増加させる動作モード(音質モード)を設け、 こ れら騒音モー ドおよび音質モー ドのうちのいずれか 1 つのモー ドを 選択的に設定できる構成を備えている。 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.
そして、 騒音モードおよび音質モー ドは、 それぞれ複数の信号特 性の所定の特性値の組合せにより規定されるものであり、 この複数 の信号特性としては、 例えば、 周波数特性、 ノイズ抑制度、 音量、 エコー除去度および側音追加量の中から選択される。 例えば、 騒音 モードの設定時には、 低域周波数帯域および高域周波数帯域が強調 されるように、 また音質モードの設定時には、 中域周波数の狭帯域 で更にその中の高域が強調されるか、 もしく は全周波数帯域が平坦 化されるように、 周波数特性が調整される。 本発明の通信端末は、 複数の種類の信号特性のそれぞれを調整するための複数個の信号処 理ユニッ トを含むデジタル処理部を備えており、 複数の信号特性の それぞれについての予め定めた特性値の組合せであって、 騒音モ一 ドもしくは音質モードに対応する特性パターンを選択的に提供でき るように、 デジタル信号処理部を制御するようになっている。 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.
複数個の信号処理ユニッ トとしては、 デジタルイコライザ、 ノィ ズ抑制回路、 ボリューム調整回路、 エコー除去回路および側音追加 回路等が用いられる。 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.
本発明のある局面によれば、 信号処理ユニッ トを、 予め定めた第
1 の特性パターンを提供するよう動作させる動作モー ド (明瞭モー ド)と、同処理ュニッ トを予め定めた第 2の特性パターンを提供する よう動作させる動作モード(自然モー ド)のうちのいずれかを選択的 に設定できるようになつており、 明瞭モードでは、 中域周波数の狭 帯域で高域を強調する周波数特性を与え、ノイズ抑制機能を増強し、 更に側音を小さく し、また自然モー ドでは、周波数特性を平坦化し、 ノィズ抑制効果を減少し、 更に側音を適度に与えるような信号特性 値が選択される。 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.
さ らに、 第 1 の特性パターンと、 第 2の特性パターンの中間的な 特性パターンを与える第 3の特性パターンを提供するよう該信号処 理ュニッ トを動作させる明瞭モードと自然モードの中間に位置する 動作モードを設定できるようにもなつている。 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.
本発明の別の局面によれば、 騒音下において、 通話音の品質を最 適にするよう、 上記複数の信号特性の各特性値を選択するよう構成 され、 受話に対しては、 低周波数帯域を強調するよう周波数特性を 調整するとともに、 側音を追加し、 送話に対してはノイズ抑制効果 を増強するとともに、 音声の中心部分の周波数帯域を残して他を抑 圧するよう信号特性値を選択するようになつている。 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.
本発明の通信端末においては、 複数の信号特性の各々について、 特定の特性値を規定する調整デ一夕を予めメモリ に記憶し、 動作モ ードの設定毎にこのメモリから読み出すようになつている。 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.
なお、 各動作モー ドの設定は、 キ一又はスィ ッチ操作で行われる とともに、 どの動作モードが設定されているかを表示器にて表示す るようになっている。
図面の簡単な説明 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
第 1 図は、 本発明の実施の形態に係る携帯電話機を示すブロック 図である。 FIG. 1 is a block diagram showing a mobile phone according to an embodiment of the present invention.
第 2図は、 本発明の第 1 の実施形態を説明するためのフローチヤ 一卜である。 FIG. 2 is a flowchart for explaining the first embodiment of the present invention.
第 3図は、 本発明の第 1 の実施形態における終話時の動作を説明 するためのフローチヤ一 トである。 FIG. 3 is a flowchart for explaining the operation at the end of a call in the first embodiment of the present invention.
第 4図は、 本発明の第 2の実施形態を説明するためのフローチヤ 一卜である。 FIG. 4 is a flowchart for explaining a second embodiment of the present invention.
第 5図は、 本発明の第 2の実施形態における終話時の動作を説明 するためのフローチヤ一トである。 FIG. 5 is a flowchart for explaining the operation at the end of a call in the second embodiment of the present invention.
第 6図は、 本発明の第 3の実施形態を説明するためのフローチヤ — 卜ャある。 FIG. 6 is a flowchart for explaining a third embodiment of the present invention.
発明の実施の形態 Embodiment of the Invention
以下図面を参照して本発明による通信端末の実施の形態を詳細に 説明する。 Hereinafter, embodiments of a communication terminal according to the present invention will be described in detail with reference to the drawings.
〈第 1 の実施の形態〉 <First embodiment>
第 1 の実施の形態は、 デジタル信号処理を用い、 使用者が受話音 声に不満があるときに、 簡単なポタン操作により ノイズ抑制、 周波 数特性、 音量、 側音、 エコー除去を一括して設定することにより、 受話を使用者が満足できる状態にすることを目的とする。 また、 そ れらの処理を一括して、 デジタル処理部に設けることにより設定を 容易にする。 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.
現在電話による通話は様々な音質になっている。 理由としては、 ( 1 )サービスの拡大により様々な事業者や通話システムが混在し、
それぞれに特有の音質を持っている。 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 ) つのシステムに特定しても様々な端末メーカにより音質が 異なる。 (2) Even if it is specified for one system, the sound quality differs depending on various terminal manufacturers.
( 3 ) 通話品質は、 使用者の年齢、 性別、 好み等により分かれる。 等が挙げられる。 (3) Call quality depends on the user's age, gender, preferences, and other factors. And the like.
これらを解決する場合、 音質について述べるときに重要視される 2つの考え方を導入することによりかなりの点を改善することがで きる。 音質を述べる場合、 会話の内容を重視し、 その明瞭さや認識 できる度合いを示す明瞭度と、 より 自然に近い細かい音まで再現さ せた自然度とを上げることが音質の向上につながる。 しかし、 これ ら 2つの特質は相反する性質をもつ。 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.
明瞭度を上げる場合 To increase clarity
( 1 ) 周波数特性は通話帯域になるべく近付け、 低域や高域のノィ ズをカツ 卜する。 (1) The frequency characteristics should be as close as possible to the communication band, and low and high frequency noise should be cut.
( 2 ) 通話帯域の 2 k H z付近を強調し、 認識度を上げる。 (2) Enhance the vicinity of 2 kHz in the speech band to increase the recognition level.
( 3 ) ノイズ抑制機能を強く し、 相手側から送られる会話以外の成 分を抑制する。 (3) Strengthen the noise suppression function to suppress components other than conversations sent from the other party.
( 4 ) 側音は小さくするか完全に無く し、 スピーカからは相手の声 のみ出力されるようにする。 (4) Sidetones should be reduced or completely eliminated, and only the other party's voice will be output from the speaker.
自然度を上げる場合 When increasing naturalness
( 1 ) 周波数特性はどの周波数でも平坦に近い状態にする。 (1) The frequency characteristics are almost flat at any frequency.
( 2 ) ノイズ抑制は少なくするか完全に無く し、 どのような音でも 細部まで聞き取りやすくする。 (2) Minimize or eliminate noise suppression to make any sound more audible to details.
( 3 ) 側音を適度な音量に設定し、 普通に会話している状態に近く する。
上記 2つの条件を満たすためにはそれぞれの設定を必要とし、 ま た単に周波数特性を変えることだけでは満たしきれない。 そこで、 これらの条件を満たすために、 デジタルイコライザ、 ノイズ抑制回 路、 ボリューム、 エコー除去回路、 および側音追加回路のような複 数の信号処理ユニッ トを一度に調整することによ り、 音声のもつ 様々な特性を再現できる。 またこれらの設定と通常携帯電話機等に 用いられるこれら 2つの設定の中間的な設定を用意し、 通話中に使 用者がこの 3種類の設定を選択できるようにすることで、 通話品質 を大幅に向上させることができる。 またこれらの設定により端末が 発生させるエコー量が増えた場合、 エコー除去回路の設定もする。 (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.
上記の第 1 の実施形態は第 1 図の携帯電話機で実現される。 すな わち、 デジタルイコライザ 1 5、 ノイズサブレッサ 1 6、 ポリユ ー ム 1 7、 エコーキャンセラ 1 8、 側音追加回路 1 9 を演算処理装置 2 4により制御させ、 制御するための調整データはメモリ 2 5 に記 憶しておく。 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.
メモリ 2 5内部の調整デ一夕としては、 イコライザ 1 5の調整パ ラメ一夕、 ノイズサブレッサ 1 6の抑圧調整データ、 ボリュームデ 一夕、 エコーキャンセラデータ、 側音追加回路 1 9の調整データ等 を用意する。 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.
使用者は、 第 2図のフローチャー トに示すように、 ステップ S 2 1 の通話中、 音質に不満等を感じ設定を変更する場合、 キー入力部 2 7 の所定のキーを操作する (ステップ S 2 2 ) 。 すると、 キ一を 押下するたびに音質の設定が切り替わる。 すなわち、 キーを押下す るたびにメモリ 2 5から「明瞭」、 「自然」、 「通常」(順不同)の音質調 整データが順次切り換わって読み出され (ステップ S 2 3 )、 この音
質調整データで演算処理装置 2 4により D S P 1 3が制御され(ス テツプ S 2 4 )、 音質の設定が変更される。 同時に L C D表示器 2 9 の表示を切り換えて (ステップ S 2 5 )、 設定された音質モードが表 示される。 この表示は、 "明瞭モー ド、 自然モード、 通常モー ド" のような使用者に分かりやすいものが適当である。 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".
終話時は、 常に通常の調整データで D S P 1 3 を制御させる。 す なわち第 3図に示すように、終了要求が出ると (ステップ S 3 1 )、 通常用調整デ一夕をメモリ 2 5から読み出し (ステップ S 3 2 )、 そ のデ一夕で演算処理装置 2 4により D S P 1 3 を制御する (ステツ プ S 3 3 )。 その後、 ステップ S 3 4の終話処理となる。 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.
なお、 上記の例では側音追加部やボリューム調整をすベて D S P 1 3の内部で行ったが、 これらは他の回路を用いても良い。 またィ コライザも他の回路で実現することが可能である。 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.
そして、 上記のような第 1 の実施形態によれば次のような効果を 得ることができる。 According to the above-described first embodiment, the following effects can be obtained.
( 1 ) 通話中、 相手の電話機の音質特性や相手の声質により通話音 質に不満を感じたときに使用者が簡単な操作により音質を好きな状 態に選べる。 (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 ) 明瞭度を上げた設定と自然度を上げた設定と、 その中間の通 常用いられる設定とを設けることで、 様々な音質に変更できる。 (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 ) 会話を特に重視したい通話の場合明瞭度を上げた設定を使用 することにより会話がスムーズになる。 (3) For a call that emphasizes the conversation, the conversation becomes smoother by using the setting with higher clarity.
( 4 ) 音楽番組等の情報ダイヤル等で様々な音を忠実に聞きたい場 合は、 自然度を上げた設定を使用することにより、 音の細部まで聞 く ことができる。
〈第 2の実施の形態〉 (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>
第 2の実施の形態は、 デジタル信号処理を用い、 使用者が騒音下 にいる場合、 簡単なボタン操作により ノイズ抑制、 周波数特性、 音 量、 側音、 エコー除去を一括して設定することにより、 送受話をノ ィズ環境下で使用するのに最適な状態にすることを目的とする。 ま た、 それらの処理を一括してデジタル信号処理部に設けることによ り設定を容易にする。 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.
ノイズ環境下で通話をする場合、 心理的音響効果により以下の現 象が起きることが想定される。 When talking in a noisy environment, it is assumed that the following phenomena will occur due to psychological acoustic effects.
受話 (自分の聞いている音) 側 Listening (the sound you are listening to)
(受 1 ) 騒音によるマスキング効果により相手の受話音が聞き取れ なくなる。 (Reception 1) The other party's received sound cannot be heard due to the masking effect of the noise.
(受 2 ) 騒音によるマスキング効果により 自分の声がマスクされ発 声音量が上がる。 (Reception 2) Your voice is masked by the masking effect of the noise, and the utterance volume increases.
(受 3 ) 何度も聞き直しているうちに大声を出し、 片耳をふさいだ りマイクに手をかざしたりする。 (Receiver 3) Speak loudly while listening repeatedly, cover one ear or hold your hand over the microphone.
送話 (相手の聞いている音) 側 Transmission (sound heard by the other party)
(送 1 ) 相手の声が大きすぎ割れている。 (Send 1) The other party's voice is too loud.
(送 2 ) 相手側の周囲ノイズが大きく耳が痛い。 (Send 2) The surrounding noise of the other party is large and my ear hurts.
(送 3 ) 相手の声が大きいので心理的効果で小さい声で対応。 相手 が聞き直すと大声で対応。 (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.
上記受 1 を解決するには周囲雑音の平均値より数 d B大きな受話 音量を必要とする。 また、 ノイズネスが強い音の高域成分とマスキ ングを受けやすい低域部分を増強し明瞭度を上げることも必要であ る。
受 2は自分の声を受話器から戻す側音を追加することにより解決 される。 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.
上記 2点を解決することにより受 3 も解決される。 Resolving the above two points will also solve the problem.
送 1 の問題は受話側を解決することにより無くなる。 送 2は送話 のノィズ抑制を強めに設定し、 且つ送話周波数特性を声の主成分で ある 5 0 0 H z〜 2 k H z にしぼり、 他の帯域を抑えるバン ドパス フィル夕を適用することで解決される。 ボリュームを若干小さくす るのも効果がある。 送 3 はこれらの問題が解決されれば無くなる。 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.
しかし、 受話音量を上げたことにより新たにエコーが発生するこ とが考えられる。 これはエコー除去機能を強くかけることにより解 決できる。 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.
上記の第 2の実施形態は第 1 図の携帯電話機により実現される。 すなわち、 デジタルイコライザ 1 5、 ノイズサブレッサ 1 6、 ポリ ユーム 1 7、 エコーキャンセラ 1 8、 側音追加回路 1 9 を演算処理 装置 2 4により制御させ、 制御するための調整データはメモリ 2 5 に記憶しておく。 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.
メモリ 2 5内部の調整データとしては、 イコライザ 1 5の調整パ ラメ一夕、 ノイズサブレッサ 1 6の抑圧調整デ一夕、 ポリュ一ムデ —夕、 エコーキャンセラデータ、 側音追加回路 1 9の調整データ等 を用意する。 これらのデータで送話及び受話を上述のように制御す ることにより、 騒音下での最適動作状態とする。 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.
使用者は第 4図に示すフローチャートのステップ S 4 1 の通話中 に騒音下での最適動作状態 (以下騒音モードと呼ぶ) を選択しょう とする場合、 ステップ S 4 2で示すようにキー入力部 2 7の選択キ 一を操作して騒音モードを選択する。 すると、 メモリ 2 5から騒音
用の調整データが読み出され (ステップ S 4 3 ) 、 このデータで演 算処理装置 2 4により D S P 1 3が制御され (ステップ S 4 9 ) 、 騒音モー ドとなる。 同時に、 L C D表示器 2 9の表示を切り換えて (ステップ S 4 5 ) 、 騒音モー ドであることを表示する。 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.
騒音モードを解除する場合は、 使用者はキー入力部 2 7の解除キ 一を操作する (ステップ S 4 6 ) 。 すると、 メモリ 2 5から通常の 調整データが読み出され (ステップ S 4 7 ) 、 このデータで演算処 理装置 2 4により D S P 1 3が制御され (ステップ S 4 8 ) 、 通常 モードとなる。 同時に L C D表示器 2 9の表示を切り換えて (ステ ップ S 4 9 )、騒音モー ドが解除された状態であることを表示する。 終話時は常に通常用のデ一夕で D S P 1 3 を制御させる。 すなわ ち、 第 5図のステップ S 5 1で終了要求が出ると、 メモリ 2 5から 通常用調整データを読み出し (ステップ S 5 2 ) 、 このデータで D S P 1 3 を制御する (ステップ S 5 3 ) 。 その後、 ステップ S 5 4 の終話処理となる。 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.
なお、 この例でも、 側音追加部やボリューム調整をすベて D S P 1 3の内部で行ったが、 これらは他の回路を用いても良い。 また、 イコライザも他の回路で実現することが可能である。 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.
そして、 上記の第 2の実施形態によれば次のような効果が得られ る。 According to the second embodiment, the following effects can be obtained.
( 1 ) 駅のホームや混雑した交差点等の周囲が騒がしいときに選択 することにより、 自分さらには相手の会話がスムースになり、 片耳 をふさいだり大声を出したりすることが無くなる。 (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 ) 相手回線によっては小さすぎる声を補正する効果もある。 なお、 上記の実施の形態はいずれも本発明を携帯電話機や P H S
(パーソナルハンディ フォンシステム) を含む移動体通信端末に応 用した場合であるが、 本発明は業務用及び趣味用の無線機などにも 利用できる。 さらに本発明は固定電話機、 公衆電話機等にも利用で さる。 (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.
〈第 3の実施の形態〉 <Third embodiment>
第 3の実施の形態は、 通話中に使用者が感じる相手の回線および 電話機の特性や相手の声質から通話をしにく いという音質的な不満 と、 自分の置かれた環境が騒音状態等の理由で、 相手の声を聞き取 ることができない音量的な不満を各々解消することを目的とする。 音質的な不満は、 相手の声がこもっていたり相手の通話回線や環 境等の理由により通話内容が理解しにくい場合や、 その逆に音楽サ —ビス等の音を聞く場合に音の細部が再生されていない場合に発生 する。 前者(音声の場合)は声の成分を強調してそれ以外のノイズ成 分等は弱く した設定を使用すればよく、 周波数特性を狭帯域で更に 高音強調形にしてノイズ抑圧を強くかけることなどで実現できる。 後者(音楽の場合)はすべての音を自然に聞き取れるようにする設定 を用意することにより解消され、 周波数特性を広帯域平坦形にして ノィズ抑圧を弱くするか全く抑圧しないようにすることで実現でき る。 この設定を "音質モード" と呼ぶ。 さらに、 前者の設定を "音 質モー ド 1 " と呼び、 後者の設定を "音質モー ド 2 " と呼ぶ。 音量 的な不満は、 自分の置かれた環境が騒音状態である場合等に発生す ることが多く、 受話音量を特大にして周波数帯域を高低域強調形に し、 且つ側音を追加して自分の声も聞き取ることができるようにす ることにより解消される。 また、 送話音も周波数帯域を狭帯域にし てノイズ抑圧を強めにすることにより相手に送られる音声を補正で
き通話がスムースになされる。 この設定を "騒音モード" と呼ぶ。 上記 2つのモー ド、 即ち 「音質モー ド」 および 「騒音モー ド」 は、 使用される状況を考えると移行操作を異ならせた方が使いやすい。 音質モードは受話音を聞きながら設定を変更できた方がよく、 これ は、 1つのキーを操作することにより音質が切り換わるようにする。 また、騒音モー ドは音量の変化が激しいため簡単に設定でき、且つ、 誤動作をしないようにする必要がある。 このことを考え、 通常、 通 話中に用いられるポリューム上昇キーの長押しで騒音モードを設定 できるようにして、 解除はボリ ューム下降キーを押すことで簡単に 解除でき、 設定前の音量に戻るようにする。 また、 イヤホン使用時 やハンズフリ一使用時は急激な音量変化による事故が発生する恐れ があるため、 設定不可とする。 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.
周囲雑音が大きい場合、 細かい音質まで聞き取ることができず、 音質的不満はおこ りにくいと考えられることより、 騒音モード時は 音質モー ドが機能しないこととして、 騒音モー ド解除時に前の状態 に戻るようにする。 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.
また、 通話開始時は通常の状態 (どちらのモードでもない状態) で始まることとし、 通話中に設定が変更されても終話時に通常の状 態に戻るようにする。 これは、 各々のモー ドが通話中に不満を感じ たときに使用することを前提としたためであり、 次回通話時に前の 設定が残っているとそれを戻す操作が必要であり、 この操作を省略 するためである。 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.
また、 通話中に F A Xやパソコン通信をしてまた通話に戻る場合 は、 前の状態を保持したままとする。 Also, if you return to the call after FAX or PC communication during a call, keep the previous state.
なお、 特定の電話番号に特定の音質調整が必要な場合も多いと考
えられる。 即ち、 音楽情報番組や特定の人物しか出ない電話番号に は、 その番号ごとに音質調整デ一夕を記憶しておいた方が使い勝手 がよく、 これを実現する場合、 電話帳のメモリ等に終話時の状態を 記憶し次回通話時にはその状態から通話できるようにすればよい。 第 1 図は上記第 3の実施形態を実現した携帯電話機を示す回路図 である。 アンテナ 1 1 により受信された電波は R F回路 1 2 に供給 され、 周波数選択等がなされる。 この出力信号は D S P (デジタル シグナルプロセッサ) 1 3の変復調回路及びモデム内に供給され、 デジタル信号に変換され音声符号変換される。 次いで、 複数個の信 号処理ユニッ ト、 即ち、 デジタルイコライザ 1 5、 ノイズ抑制回路 (ノイズサブレッサ) 1 6、 ボリューム 1 7、 エコー除去回路 (ェ コ一キャンセラ) 1 8、 側音追加回路 1 9を順次経る。 そして、 D / A変換器 2 0でアナログ信号に変換されスピーカ 2 1 により音声 に変わる。 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.
一方、 マイク 2 2から入力された音声は A Z D変換器 2 3でデジ タル信号に変換され逆の手順によりアンテナ 1 1で電波として送信 される。 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.
この携帯電話機ではデジタルイコライザ 1 5、 ノイズサブレッサ 1 6、 ボリューム 1 7、 エコーキャンセラ 1 8、 側音追加回路 1 9 が演算処理装置 2 4により制御され、 制御するための調整データは メモリ 2 5に記憶しておく。 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.
メモリ 2 5内部の調整データとしては、 イコライザ 1 5の調整パ ラメ一夕、 ノイズサブレッサ 1 6の抑圧調整データ、 ボリュームデ —夕、 エコーキャンセラデータ、 側音追加回路 1 9の調整デ一夕等 を用意する。
上述した音質、 騒音の各モー ドでは上記メモリ 2 5 に格納する 各々の回路の制御デ一夕を一まとまりにしてその組み合わせで各モ ードを実現する。 下記表 1はその組み合わせパターンである。 通常 の状態の受話デ一夕 R (0 )と送話データ T (0 )をデフォルトとして 各モー ド設定時、受話側はすべての状態で別のデータ R ( 1 ), R (2 R (3 )を使用し、 送話側は騒音モ一 ドのみ設定値をデータ T ( 1 )に 変える。 音質モー ドは、 前述した音質モー ド 1、 音質モード 2に対 応して通常以外に 2つを想定したがこれ以上あっても構わない。 な お、 第 1図において、 2 6はプログラムを記憶する R OM、 2 7は 各種入力及び設定を行うキー入力部、 2 9は電話番号や設定状態を 表示する L C D表示器である。 L C D表示器 2 9は、 演算処理装置 2 4により L C D ドライバ 2 8を介して制御される。 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.
第 6図は、 上記携帯電話機で第 1の実施形態の動作が行われたと きのフローチャー トである。 このフローチャー トを説明すれば、 ス テツプ S 1での通話中 (通話開始時は受話デ一夕及び送話データが 表 1の R (0 ), T (0 )で標準モードである) にイヤホン使用中また はハンズフリー動作中かが判断され (ステップ S 2 ) 、 もしそうで あれば騒音モー ドへの変更を禁止してステップ S 3に進む。そして、 ステップ S 3でキーが押下され、 音質モー ドへの変更が要求される
と、 ステップ S 4に進んでキー押下毎に受話データを R (l )または R (2 )または R (0 )に切り換え、 モー ドを音質モー ド 1 または音質 モード 2または標準モードに切り換える。 同時に、 切り換えられた モード (音質モード 1 または音質モード 2または標準モード) を L C D表示器 2 9に表示させる。 そして、 音質モード 1 または音質モ ー ド 2または標準モー ドでの通話中に終話キーが押されると (ステ ップ S 5 ) 、 ステップ S 6の終話処理となり、 モードが音質モー ド 1 または 2の場合は受話デ一夕及び送話データが R (0 ), T (0 )に 戻され標準モードに戻る。 一方, ステップ S 3で音質モードへの変 更が要求されない場合は、 標準モードのまま通話を継続してステツ プ S 5で終話キーが押下されれば、ステツプ S 6の終話処理となる。 ステップ S 1 による通話中、 またはステップ S 5で終話キーが押 下されないことによる通話中に、 ステップ S 2でイヤホン使用中ま たはハンズフリ一動作中かを判別し、 もしそうでなければステツプ S 7に進んで騒音モー ドへの変更を可能にする。 そして、 ステップ S 7でボリューム上昇キ一が長押しされると、 ステップ S 8に進ん で現在の音質モー ド状態を示す受話データをメモリ 2 5に記憶し、 且つ騒音モー ドになることを L C D表示器 2 9に表示した上で、 受 話デ一夕を R (3 )、送話デ一夕を T ( l )とし、騒音モードとする (ス テツプ S 9 ) (したがってステップ S 4で音質モード 1 または 2に設 定されていても、この音質モードはここで一旦終了する)。しかる後、 ステップ S 1 0でボリューム下降キーが押下されると、 またはィャ ホン使用、 あるいはハンズフリー状態になると、 ステップ S 1 1 に 進み送話データを Τ (0 )、 受話デ一夕を記憶していた以前のデ一夕 に戻して、 騒音モード以前のモー ドに戻り、 ステップ S 3に進む。
ステップ S 1 0でボリューム下降キーが押下されず、 またはィャホ ン非使用状態、 あるいはハンズフリー動作状態にないとき、 ステツ プ S 1 2 に進み、 終話キーが押下されると (ステップ S 1 2 ) 前述 したステップ S 6 の終話処理を行う。 ステップ S 7でポリューム上 昇キーが長押しされず、 騒音モードに進まない場合もステップ S 3 に進む。 そして、 ステップ S 3以後は前記と同様である。 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.
なお、 上記のキー操作に代えて、 回転式スィ ッチを使用して、 回 すと音質モー ド、 長押しで騒音モー ド、 あるいはシ一ソースイ ッチ を使用して、 上下で音質モー ド、 長押しで騒音モー ドとすることも できる。 さらに、 騒音モードの解除に回転式スィ ッチまたはシーソ —スィ ッチを使用することもできる。 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.
そして、 上記の第 3の実施形態によれば、 携帯電話機での通話中 に不満に思われる相手側の電話機や環境、声質による音質的不満と、 自分が周囲雑音の多い場所等にいる理由から発生する音量的不満を 解消できる。 しかも、 解消する状態に電話機をスムースに且つ簡単 に設定することができる。 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.
産業上の利用可能性 Industrial applicability
以上詳細に説明したように本発明の通信端末によれば、 相手の声 質、 回線や相手端末の特性、 および自分の置かれた環境等から発生 する各種の問題を除去して良好な通話品質を得ることができ、 通話 が快適となる。
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
1 . 請求項 1 0 に記載の通信端末において、 1. The communication terminal according to claim 10,
受話に対しては、 低域周波数および高域周波数を強調するよう 周波数特性を調整するとともに、 側音を追加し、 送話に対しては ノイズ抑制効果を増強するとともに、 音声の成分の周波数帯域を 残す特性値を選択するようにした通信端末。 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 . 請求項 5 に記載の通信端末において、 2. The communication terminal according to claim 5,
該複数の信号特性の各々について、 特定の特性値を規定する音 質調整データを予めメモリに記憶し、 動作モー ドの設定の度にこ のメモリから読み出すようにした通信端末。 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 . 請求項 1 2 に記載の通信端末において、 3. The communication terminal according to claim 12,
該音質調整データを相手電話番号と関連づけて該メモリ に記 憶し、 次回の通話時に該メモリから読み出すようにした通信端末 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 . 請求項 1又は 5 に記載の通信端末において、 4. The communication terminal according to claim 1 or 5,
各動作モードの設定が、 キー又はスィ ッチ操作で行われる通信 顺末。 Communication terminal where each operation mode is set by key operation or switch operation.
5 . 請求項 1又は 5 に記載の通信端末において、 5. The communication terminal according to claim 1 or 5,
どの動作モー ドが設定されているかを表示器に表示する通信 端末。 Communication terminal that displays which operation mode is set on the display unit.
6 . 請求項 1 4に記載の通信端末において、 6. The communication terminal according to claim 14,
各動作モードが、 独立のキー又はスィ ッチ操作により別の動作 モー ドに切り換えられる通信端末。 A communication terminal in which each operation mode can be switched to another operation mode by an independent key or switch operation.
7 . 請求項 1 4に記載の通信端末において、 7. The communication terminal according to claim 14,
キー又はスィ ッチの長押しによ り所定の動作モー ドに移行さ れる通信端末。
A communication terminal that shifts to a predetermined operation mode when a key or switch is pressed for a long time.
8 . 請求項 1 4に記載の通信端末において、 8. The communication terminal according to claim 14,
キー又はスィ ッチで特定の動作モー ドを解除し、 音量を元の状 態に戻すようになつている通信端末。 A communication terminal that releases a specific operation mode with a key or switch and returns the volume to its original state.
9 . 請求項 1 4に記載の通信端末において、 9. The communication terminal according to claim 14,
特定の動作モー ドが、 単一のキ一又はスィ ッチ操作で行なわれ る通信 ¾末。 A communication terminal in which a particular mode of operation is performed with a single key or switch operation.
0 . 請求項 1又は 5 に記載の通信端末において、 0. In the communication terminal according to claim 1 or 5,
特定のモード選択時は、 他のモー ドの選択を禁止させて、 受け つけないようにする通信端末。 A communication terminal that, when a specific mode is selected, prohibits other modes from being selected and does not accept them.
1 . 請求項 1 、 5又は 1 0 に記載の通信端末において、 1. The communication terminal according to claim 1, 5, or 10,
イヤホン使用時やハンズフリー機能の使用時には、 該第 1 の動 作モードを受けつけず、 該第 1 の動作モード中にイヤホン使用や ハンズフリー機能の使用が発生した場合は、 第 1 の動作モー ドを 解除するようになっている通信端末。 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 . 請求項 1又は 5 に記載の通信端末において、 2. The communication terminal according to claim 1 or 5,
終話時には通常設定(デフォルト)に戻るようにし、 次回通話時 は該デフォルトで開始される通信端末。 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 . 請求項 2 2 に記載の通信端末において、 3. The communication terminal according to claim 22,
該第 2の動作モー ドに移行し、 設定された後において、 データ 通信モードに移行後に再び通話モー ドへ移行する時は、 デフオル 卜に戻さないようにする通信端末。
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.
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
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JP31167699A JP3769433B2 (en) | 1999-11-01 | 1999-11-01 | Communication terminal |
JP11/311676 | 1999-11-01 |
Publications (1)
Publication Number | Publication Date |
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WO2001033813A1 true WO2001033813A1 (en) | 2001-05-10 |
Family
ID=18020140
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
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PCT/JP2000/007665 WO2001033813A1 (en) | 1999-11-01 | 2000-10-31 | Communication terminal |
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JP (1) | JP3769433B2 (en) |
WO (1) | WO2001033813A1 (en) |
Families Citing this family (6)
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US7983920B2 (en) * | 2003-11-18 | 2011-07-19 | Microsoft Corporation | Adaptive computing environment |
JP4928922B2 (en) | 2006-12-01 | 2012-05-09 | 株式会社東芝 | Information processing apparatus and program |
JP5034595B2 (en) * | 2007-03-27 | 2012-09-26 | ソニー株式会社 | Sound reproduction apparatus and sound reproduction method |
JP2010171839A (en) * | 2009-01-26 | 2010-08-05 | Furuno Electric Co Ltd | Wireless communication set |
JP6168451B2 (en) * | 2013-07-11 | 2017-07-26 | パナソニックIpマネジメント株式会社 | Volume adjustment device, volume adjustment method, and volume adjustment system |
US20230169989A1 (en) * | 2020-04-02 | 2023-06-01 | Dolby Laboratories Licensing Corporation | Systems and methods for enhancing audio in varied environments |
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JPH0383437A (en) * | 1989-08-28 | 1991-04-09 | Sharp Corp | Cordless telephone set |
US5384843A (en) * | 1992-09-18 | 1995-01-24 | Fujitsu Limited | Hands-free telephone set |
JPH06209362A (en) * | 1993-01-11 | 1994-07-26 | Matsushita Electric Ind Co Ltd | Anti-sidetone speaking device |
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JP2001136239A (en) | 2001-05-18 |
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