JP2629618B2 - Gain adjustment device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a gain adjusting device,
In particular, the present invention relates to gain adjustment used in a receiver of a mobile telephone.

[0002]

2. Description of the Related Art Conventionally, a receiver of this type of mobile telephone has
It is used for the purpose of demodulating and amplifying a received signal and outputting a sound of a volume set by a carrier.

FIG. 3 is a block diagram of a receiver section of a conventional mobile telephone.

[0004] In the figure, a receiving unit includes an antenna 1, a radio unit 2, a digital signal processor 30 (hereinafter referred to as DSP),
SP control unit 70, digital / analog conversion unit (hereinafter D
/ A conversion unit) 8, receiving volume controller 9, receiver amplifier 1
0, a receiver 11.

[0005] The DSP 30 is provided with a channel decoder 40.
And a speech decoder 50.

Next, the operation will be described.

The digital data received via the antenna 1 and demodulated by the radio unit 2 has a predetermined time width (for example, 9
0 msec) and audio data for each frame and control data. Demodulated data from the radio unit 2 is sent to a channel decoder 40 in the DSP 30
The data is demodulated into data for each frame, and the voice data in the frame is further demodulated in the speech decoder 50 to obtain digital voice data. The digital audio data obtained by the speech decoder 50 is converted into an analog signal by the D / A converter 8, the level of which is manually adjusted by the reception volume controller 9, and amplified by the receiver amplifier 10, and amplified by the receiver amplifier 10. The audio is output from the receiver 11 such as a speaker. The DSP controller 70 controls the DSP 30 in a system.

In the receiving section of such a conventional portable telephone, the level of the output sound is adjusted by the receiving volume controller 9.
Had to be carried out directly by hand.

FIG. 4 is a block diagram of a receiver function of a hands-free automobile telephone having a volume control means described in Japanese Patent Application Laid-Open No. 3-123130 published May 24, 1991.

In FIG. 1, an automobile telephone device 18 includes a handset 17, a microphone 16 used during a hands-free call, a speaker 15, a volume control means 13 for controlling a speaker volume, and a telephone control means 12 for controlling these. Consists of The mobile telephone radio unit 19 connects a base station (not shown) and the mobile telephone device 18 via a wireless line.

Next, the operation will be described.

During a call, a signal corresponding to the traveling speed of the vehicle is output from the speed detecting means 14 to the volume control means 13, and the volume of the speaker 15 is controlled by this signal. For this reason, the driver can make a call without adjusting the receiving sound volume in response to the change in the ambient noise caused by the change in the traveling speed.

[0013]

However, in the receiver section of the conventional mobile telephone shown in FIG. 3, variations in the gain of the radio line to the base station and the connection line connecting each circuit, and the opposite terminal, Of the receiving level becomes very large due to the variation of the transmitting sensitivity of the receiving side and the variation of the utterance volume by the generator on the opposite terminal side, and cannot be covered by the receiving volume controller 9 alone. was there.

Further, even in the receiving section of the conventional hands-free mobile telephone shown in FIG. 4, even if an automatic adjusting function of the receiving sound volume corresponding to the running speed of the automobile and the amount of ambient noise is configured, as described above. There is a problem that there is no effect on a change in the reception level due to a complicated line and an opposite terminal.

An object of the present invention is to solve the above-mentioned problems,
At the optimal volume level, the audio signal included in the received signal is
An object of the present invention is to provide a gain adjustment device capable of automatically demodulating and outputting.

[0016]

Means for Solving the Problems In order to achieve the above-mentioned object, a gain adjusting device according to the present invention has a predetermined configuration.
Audio signal with a fixed frame length and the control signal
An antenna for receiving a wireless modulated digital signal,
Demodulates a wireless modulated digital signal and outputs a digital demodulated signal.
And a digital unit for transmitting the digital demodulated signal for each frame.
And a channel decoder that divides the
Audio signal is included in the divided digital demodulated signal
A determination unit for determining whether or not the audio signal has been determined
When the frame is divided for each frame for a predetermined time,
And integrate the energy of the digital demodulated signal
A measuring unit, the integrated energy, and a predetermined value
Compares two different threshold values
The comparison section and the integrated energy are the two threads.
Higher than the higher threshold value among the threshold values
Set the step level to a predetermined negative value
When it is lower than the threshold value, the step level
Set to a predetermined positive value, the two thresholds
Set the step level to "0" when between values
A setting unit for adding the step level to the audio signal
An audio signal to which the step level is added
Digital-to-analog converter that converts digital to analog audio signals
An amplification unit for amplifying the audio analog signal;
And a speaker for outputting a widened audio analog signal .

[0017]

The gain adjusting device according to the present invention employs the above-described configuration, whereby the measuring means detects the volume level of the audio signal by integrating the energy of the digital input signal, and the comparing means, the setting means, and the adding means. The means reduces the volume level by a predetermined amount when the audio signal has a volume level equal to or higher than a predetermined level, and decreases the volume level by a predetermined amount when the audio signal has a volume level lower than a predetermined level. In order to raise the volume level, a sound having a constant volume is output.

[0018]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Next, the present invention will be described in detail with reference to the drawings.

FIG. 1 is a block diagram of a receiver of a portable telephone showing one embodiment of the present invention.

In FIG. 1, the receiving unit includes an antenna 1, a radio unit 2, a DSP controller 7, a D / A converter 8, a receiving volume controller 9, a receiver amplifier 10, and a receiver 11.
These components are the same as the conventional one. Also,
The DSP 3 includes a channel decoder 4 having the same function as that of the related art,
A speech decoder 5 in which functions of signal regression analysis, voice signal determination, frame energy measurement, and frame energy integration are added to a conventional speech decoder, and a receiving interface based on the result of frame energy integration performed by the speech decoder 5 And a gain controller 6 for automatically adjusting the level to an appropriate level. D
SP3 is controlled by a DSP controller 7 equipped with DSP control software.

Next, the operation of the gain adjusting device according to the present invention will be described in detail with reference to the flowchart for explaining the operation of the DSP 3 shown in FIG.

First, in FIG. 1, the antenna 1 receives a radio signal transmitted from a base station (not shown), and the radio unit 2 demodulates the received radio signal to output a digital demodulated signal. The digital demodulated signal is input to a channel decoder 4 constituting the DSP 3, and a digital demodulated signal for each frame is output to a speech decoder 5.

Hereinafter, description will be made with reference to the flowchart of FIG.

After the speech decoder 5 starts the speech decoding process (step S1), the speech decoder 5 analyzes the regression of the audio data for each frame transmitted from the channel decoder 4 (step S1).
2). Regression means repeatability. Since voice data is usually highly periodic, it has high regression. On the other hand, regression is low because noise is random. Therefore, whether or not the input data is voice can be determined on the basis of the degree of regression. In step S3, it is determined whether or not the speech is based on the thus obtained high regression. Here, when the input data is a howling or tone signal, the regression becomes abnormally high. Therefore, in order to distinguish these signals from speech symbols, the regression has two different levels. If the input is within the range between the threshold values, the input is determined to be audio.If a higher regression is recognized than the high threshold value, the input is converted to audio as an abnormal signal such as howling or a tone signal. Is not determined.

In step S3, if it is determined that the voice is not a voice, the process returns to step S2. If it is determined that the voice is a voice, the frame energy is measured for a predetermined number of frames, ie, a predetermined time. (Step S4). The frame energy corresponds to the volume level of the voice, and the measurement of the frame energy is usually several hundred meters.
This is performed for a plurality of frames included in a section of about sec. Next, in step S5, the frame energy obtained in step S4 is integrated. This integral is
This is a process of accumulating frame energy obtained in the past.

The integrated value of the obtained frame energy is
In step S6, the high threshold value TH
Is compared with a low threshold value TL, and when it is higher than TH, it is determined as “large”, when it is lower than TL, it is determined as “small”, and when it is between TH and TL, it is determined as “medium”. , The subsequent processing is executed.

First, when the integral value is determined to be "large",
That is, when it is determined that the volume level of the sound is “high”, the timer count-up operation is started (step S7). The set time of this timer is usually set to several seconds. The timer count-up operation by the timer is performed to determine a stable audio signal whose input is not instantaneous but continues for a relatively long time. That is, as described above, the determination as to whether or not the input data is a voice is made by analyzing the signal regression. However, the voice determination cannot be reliably performed only by the determination from the signal regression. . Therefore, when a plurality of frames, that is, a section of about several seconds, in the above-described analysis results of the signal regression continue when a speech signal is determined to be a speech signal, the speech decoder 5 recognizes the speech as speech and will be described later. Processing is performed. Therefore, in step S8, it is determined whether or not the timer set time has elapsed, that is, a time-out has occurred. If the time has elapsed, the input signal at this time is recognized as an audio signal, and the step level is set to "-1". After this (step S9), the timer is reset to prepare for the next input signal.
After this timer reset, or when it is determined in step S8 that the time has not expired,
After recognizing that the input signal is not a voice signal, step S1
Move to 6. In step S16, the set step level is added to the audio data, and the process returns to step S2. That is, if the input signal is an audio signal, the step level "-1" is added. If the input signal is not an audio signal, the step level is not added.

If it is determined in step S6 that the integral value of the frame energy is "medium", that is, if it is determined that the sound volume level is appropriate, the step level is set to "0". , (Step S
11), the process proceeds to step S16 and proceeds to step S2.
Return to the processing of.

When it is determined that the integral value of the frame energy is "low", that is, when it is determined that the volume level of the voice is "low", the timer count-up operation is performed in the same manner as in steps S7 and S8. Is started (step S12), and a timeout is determined (step S12).
13). In step S13, if the timeout has not occurred, the process proceeds to step S16. If the timeout has occurred, the input signal at this time is recognized as an audio signal, and the step level is set to "1" (step S14).
Thereafter, the timer is reset to prepare for the next input signal (step S15). After resetting the timer, the process proceeds to step S16, where the set step level is added to the audio data (step S16), and the process returns to step S2. That is, when the input signal is an audio signal, the step level “1” is added, and when the input signal is not an audio signal, the step level is not added.

In the above description, the step levels "1" and "-1" correspond to increasing and attenuating the level by 1 dB, respectively, and the level of the audio data is adjusted by performing the processing of step S16. It will be.

By repeating the above-described routine processing, the level of the received voice from the channel decoder 4 can be adjusted to a predetermined range (between the two threshold values in step S6). That is, if the received voice level is lower than TL and it is difficult to hear, the step level is set to "1" in step S14, and the voice data level is increased by 1 dB in step S16, while the received volume level is higher than TH. in case of,
In step S9, the step level is set to "-1". In step S16, the audio data level is attenuated by 1 dB, and eventually the received audio level is adjusted between the thresholds TH and TL, which are the optimal audio levels. Will be able to

In the embodiment shown in FIG. 1, the processing performed by the speech decoder 5 and the gain controller 6 constituting the DSP 3 is the same as that shown in FIG.
Corresponds to the processing shown in FIG. In this embodiment as well, the received sound volume adjuster 9 for externally adjusting the received sound volume is provided, but it is apparent that the adjuster 9 may not be provided.

In the above embodiment, whether or not the input signal is an audio signal is analyzed based on the regression of the signal. However, whether or not the input signal is an audio signal is determined by another known method. Of course, it may be executed based on the above.

[0034]

As described above, in the gain adjusting apparatus according to the present invention, when the reception level is lower than a certain level and the level is hard to hear, or when the reception level is higher than a certain level, the level at which a sense of distortion is present is obtained. , The audio level is automatically and smoothly fixed to an appropriate level, so that the wearer can always have a comfortable call.

[Brief description of the drawings]

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

FIG. 2 is a flowchart for explaining the operation of the DSP shown in FIG. 1;

FIG. 3 is a block diagram of a receiver of a conventional mobile telephone.

FIG. 4 is a block diagram of a receiver of a conventional hands-free automobile telephone.

[Explanation of symbols]

 DESCRIPTION OF SYMBOLS 1 Antenna 2 Radio | wireless part 3 Digital signal processor (DSP) 4 Channel decoder 5 Speech decoder 6 Gain control part 7 DSP controller 8 D / A conversion part 9 Receiving sound volume adjuster 10 Receiver amplifier 11 Receiver 12 Telephone control means 13 Volume control means 14 Speed detection means 15 Speaker 16 Microphone 17 Handset 18 Car telephone device 30 DSP 40 Channel decoder 50 Speech decoder 70 DSP controller

Claims (3)

(57) [Claims] 1. It has a predetermined constant frame length.
Modulated digital signal consisting of audio signal and control signal
And an antenna for receiving the radio-modulated digital signal and a digital demodulated signal.
And a channel for dividing the digital demodulated signal for each frame.
A sound signal to the digital demodulated signal divided for each frame.
A determination unit for determining whether or not a signal is included; and a predetermined time when the voice signal is determined.
The energy of the digital demodulated signal divided for each frame
A measuring unit that integrates and measures the energy, and a difference between the integrated energy and a predetermined value.
Comparison unit for comparing two threshold values
And the integrated energy is the two thresholds
If the value is higher than the high threshold value,
Set the threshold level to a predetermined negative value to
When the step level is lower than the
Set to be between the two threshold values
A setting unit for setting a step level to “0”; and an adding unit for adding the step level to the audio signal.
And the audio signal to which the step level has been added is converted to an analog sound.
A gain adjustment device comprising: a digital-to-analog conversion unit that converts a voice signal; an amplification unit that amplifies the audio analog signal; and a speaker that outputs the amplified audio analog signal . 2. The method according to claim 1, wherein the determining unit is configured to divide the frame into frames.
Of the audio data contained in the demodulated digital demodulated signal
Analyzing to determine the presence or absence of the audio signal
2. The gain adjusting device according to claim 1, wherein: 3. An insert between the speaker and the amplifying unit.
Adjusting the output level of the audio analog audio signal.
2. The apparatus according to claim 1, further comprising a receiving volume control unit.
On-board gain adjustment device.