CN100593323C - Method and system for automatic volume adjustment - Google Patents

Abstract

The present invention relates to a method and system capable of automatic volume adjustment, specifically comprising: dividing input audio data into high and low frequency bands; calculating the average volume of the high and low frequency band data respectively; converting the average volume data to logarithmic domain, and according to the control curve Take the value to obtain the control gain data; convert the obtained data to the linear domain, and multiply them with the corresponding delayed frequency band data respectively; add and output the multiplied high and low frequency band audio data. The invention realizes the automatic adjustment of the audio volume, controls the output volume within the range specified by the user, and does not cause sound distortion. The method and system can not only be applied to television systems, but can also be extended to fields using audio signals such as mobile terminals, broadcasting systems, computers, and vehicle-mounted CDs.

Description

Method and system for automatic volume adjustment

technical field

The invention relates to a method and system for automatic volume adjustment, in particular to a method and system for automatically adjusting volume and preventing sudden changes in volume.

Background technique

Users can manually set the volume and other parameters of TV programs to a certain value, but the volume of different TV program sources may be loud or small. For example, the audio volume of different channel signal sources varies greatly; The purpose of attention is often greater than the volume of normal programs. Therefore, when the TV channel is changed or the program of a certain channel is changed, the volume often increases or decreases suddenly, which brings uncomfortable feeling to the audience.

The utility model with the patent No. 200320119709 proposes the following technical solution from the perspective of analog signal processing to solve the above problems: detect the audio volume signal at the audio input end through the detection circuit and send it to the control circuit. When the required volume is shifted, the control circuit is connected to the volume adjustment circuit; through the volume adjustment circuit, the "plus" or "minus" key is used to adjust the voltage amplitude of the accompanying sound to realize the purpose of automatic volume control.

The above technical solution realizes the automatic adjustment of the volume to a certain extent, but only through the technical solution of modifying the amplitude of the analog audio signal. Adjusting the voltage amplitude of the accompanying sound through the "plus" or "minus" key may cause the control signal to be too high. Rapid changes introduce additional noise.

At the same time, a method of automatically controlling the volume from the perspective of digital processing has also appeared on the market. The analog signal is processed through the A/D converter, and the buffer is used to calculate the root mean square of the processed audio data to obtain the input data. After the average volume of the input data is given in the logarithmic domain to control the average volume of the input data, it is output through the D/A converter. Since the above-mentioned technical solution uniformly processes the input audio, when the low-frequency component of the input signal is dominant, the high-frequency component with a small amplitude will be lost after the above-mentioned method, so that the processed audio loses its original timbre or content; using buffers to calculate RMS takes a lot of system resources.

In order to overcome the above problems, there is a need for a method and system that can automatically adjust the volume, so that the sudden change of the output volume can be controlled within a certain range without causing sound distortion. The method and system can not only be applied to television systems, but can also be extended to fields using audio signals such as mobile terminals, broadcasting systems, computers, and vehicle-mounted CDs.

Contents of the invention

The purpose of the present invention is to provide a method for automatic volume adjustment in view of the defects of the existing audio processing technology in the aspect of automatic volume adjustment. The high-frequency signal part of the output audio data will not be lost; by introducing a smoothing filter and designing a control curve, the change of the control signal tends to be smooth.

The purpose of the present invention is also to provide a system for automatic volume adjustment, which can be widely used in systems such as televisions, mobile terminals, computers, and vehicle-mounted CDs, to realize automatic volume adjustment without affecting audio effects.

To achieve the above object, the present invention provides a method for automatic volume adjustment, comprising the following steps:

Step 1, dividing the input audio data into high frequency band audio data and low frequency band audio data;

Step 2, calculating the average volume parameters of the high and low frequency band audio data respectively;

Step 3. Convert the average volume data to the logarithmic domain, and obtain the control gain data according to the value of the control curve;

Step 4, converting the obtained data into the linear domain, and multiplying them with the delayed corresponding frequency band data respectively;

Step 5, adding and outputting the high and low frequency band audio data obtained after multiplication.

The present invention also provides an automatic volume adjustment system, which receives the audio signal output by the A/D converter and sends it out after automatic volume adjustment; it includes a low-pass filter connected in sequence, a root mean square calculation module, a control module and a multiplication The low-pass filter module is connected to the multiplication module through the delay module; it also includes a high-pass filter, root mean square calculation module, control module and multiplier module connected in sequence, and the high-pass filter module is connected to the multiplication module through the delay module; The two multiplication modules are respectively connected to the adder module.

Therefore, the present invention has the following advantages:

1. Automatic volume adjustment controls the volume of the output audio within a certain range to prevent the volume from suddenly increasing or decreasing, so that the audience can obtain a comfortable hearing experience;

2. Using high-pass/low-pass filters to divide the input audio signal into two frequency bands for targeted processing, preserving the original timbre and content before processing;

3. The root mean square calculation uses a first-order wireless pulse filter with a certain time constant for calculation, which avoids the problem of excessive calculation resources required in the calculation method that usually uses a buffer;

4. The control curve adopts a soft knee control curve, and at the same time uses a smoothing filter for smoothing, which avoids the introduction of additional noise due to the rapid change of the control signal.

The technical solutions of the present invention will be described in further detail below in conjunction with the accompanying drawings and embodiments.

Description of drawings

Fig. 1 is the flowchart of audio processing method in the prior art;

Fig. 2 is the flowchart of automatic volume adjustment method of the present invention;

Fig. 3 is the schematic diagram of the method for root mean square calculation in the automatic volume processing of the present invention;

Fig. 4 is the soft knee control curve schematic diagram in the automatic volume processing of the present invention;

Fig. 5 is a system block diagram of audio processing in the prior art;

Fig. 6 is a system block diagram of automatic volume adjustment in the present invention.

Detailed ways

The usual audio processing converts the input audio data into digital audio data, controls it according to the parameters set by the user, and then converts it into analog audio data output; at the same time, in order to overcome the sudden change of the volume, before controlling the audio data according to the user setting, add A method for automatic volume adjustment. Referring to FIG. 1 , it is a flowchart of an audio processing method in the prior art.

The present invention provides a method for automatic volume adjustment aimed at the defects of existing automatic volume adjustment, as shown in Figure 2, including the following steps:

Step 101, use a high-pass/low-pass filter to divide the input audio data into high-band audio data and low-band audio data; because the frequency range of the subwoofer is usually selected from 0-200Hz, the cutoff frequency (cutoff frequency) that distinguishes the high and low frequency bands It is 200Hz, audio data below 200Hz is low-frequency audio data, and audio data above 200Hz is high-frequency audio data;

Step 102, using a first-order infinite impulse response filter with a certain time constant to calculate the root mean square (Root Mean Square, RMS for short) of the high and low frequency band data respectively, to obtain the respective average volume parameters of the high and low frequency bands;

Step 103, converting the obtained average volume data into logarithmic domain;

Step 104, obtain control gain data according to the value of the preset soft knee control curve;

Step 105, converting the logarithmic domain audio data obtained according to the control curve into a linear domain;

Step 106, using a first-order infinite impulse response filter to perform low-pass smoothing filtering;

Step 107, multiplying the filtered high and low frequency band audio data with the delayed corresponding data of this frequency band respectively;

Step 108, adding the high and low frequency band audio data obtained after multiplication;

Step 109, output.

In step 102 above, refer to FIG. 3 for a schematic diagram of RMS calculation. The transfer function is:

$\mathrm{<span\; class="notranslate">\; h</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; z</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; =</span>\frac{\mathrm{<span\; class="notranslate">\; TAV</span>}}{\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; -</span><span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; TAV</span>}<span\; class="notranslate">\; )</span>{\mathrm{<span\; class="notranslate">\; Z</span>}}^{<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 1</span>}}}$

Among them, TAV is the average coefficient, and the relationship between its value and the window length RMS_average_in_Samples used when the sliding window is used for averaging is as follows:

TAV=exp(ln(0.01)/RMS_average_in_Samples);

The RMS calculation formula is as follows:

$\mathrm{RMS}=\sqrt{\frac{x_1^2+x_2^2+\&Center\; Dot;\&Center\; Dot;\&Center\; Dot;+x_N^2}{N}},$ Where x ₁ , x ₂ , . . . , x _N are the volumes of the input audio data respectively.

In the above step 104, the value is taken according to the soft knee control curve, referring to Fig. 4, the soft knee control curve on the logarithmic domain is designed according to the hard inflection point control curve. In the hard knee curve, when the input dB value is less than -30dB, the output is the same as the input, that is, the control gain is 0dB; and when the input dB value is greater than -30dB, the output value remains at -30dB, that is, the control gain is Although the input changes; the soft knee curve is designed based on the idea of smooth transition at the corner of the hard knee. When the input value is around -30dB, the output value is selected according to the smooth curve in the figure, which is the value according to the preset soft knee control curve.

In step 105, the smoothing filter algorithm is the same as the algorithm for obtaining RMS, but the average coefficient TAV in the transfer function is not constant, and for the two cases where the control gain is on an upward trend and is on a downward trend, the TAV The values are different, the time constant is smaller for uptrends (20ms) than for downtrends (16s).

Referring to Figure 5, it is a schematic diagram of a system for audio processing in the prior art, including an A/D conversion module, an audio processing module, and a D/A conversion module connected in sequence; wherein, the audio processing module controls audio data according to user settings deal with. At the same time, in order to overcome sudden changes in volume, an automatic volume adjustment module is added before the audio processing module.

Referring to Fig. 6, it is a system block diagram of the automatic volume adjustment system in the present invention, the automatic volume adjustment module automatically adjusts the size of the input audio data RMS, controls the volume of the output audio within a certain range, and prevents the sudden increase of the volume.

The automatic volume adjustment system includes a low-pass filter 1, root mean square calculation module 31, logarithmic domain conversion module 41, dynamic control law and threshold module 51, linear domain conversion module 61, smoothing filter module 71 and multiplication The device module 81, the low-pass filter 1 is connected to the multiplication module 81 through the delay module 91; It also includes the high-pass filter 2 connected in sequence, the root mean square calculation module 32, the logarithmic domain conversion module 42, the dynamic control law and the threshold limit module 52, linear domain conversion module 62, smoothing filter module 72 and multiplier module 82, high-pass filter 2 is connected to multiplication module 82 through delay module 92; Two multiplication modules 81 and 82 are connected to adder module 10 respectively.

In this system, the low-pass filter 1 and the high-pass filter 2 are second-order wireless impulse response filters; the RMS calculation modules 31 and 32 are first-order infinite impulse response filters with a certain time constant; the dynamic control The soft knee control curve is pre-designed in the law and threshold modules 51 and 52, so that the received logarithmic domain audio data takes values according to the soft knee control curve to obtain the control gain; First-order wireless pulse response filters with different time constants for the small case.

Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention without limitation. Although the present invention has been described in detail with reference to the preferred embodiments, those of ordinary skill in the art should understand that the technical solutions of the present invention can be The scheme shall be modified or equivalently replaced without departing from the spirit and scope of the technical scheme of the present invention.

Claims (11)

1, a kind of method of automatic volume adjusting is characterized in that, comprises the steps:

Step 1, input audio data is divided into high band voice data and low-frequency range voice data;

Step 2, calculate the average volume parameter of high and low frequency section audio data respectively;

Step 3, with the average volume data transaction to log-domain, taking parameter on trendline according to control obtains the ride gain data;

Step 4, with the data transaction that obtains to linear domain, multiply each other respectively with through the corresponding frequency band data that postpone;

Step 5, the high and low frequency section audio data addition output that obtains after will multiplying each other.

2, the method for automatic volume adjusting according to claim 1, it is characterized in that, in the described step 1, the step that input audio data is divided into high band voice data and low-frequency range voice data is: use high pass/low pass filter that input audio data is divided into high band voice data and low-frequency range voice data.

3, the method for automatic volume adjusting according to claim 1, it is characterized in that, in the described step 2, the average volume parameter of calculating high and low frequency section audio data respectively comprises: use the single order infinite impulse response filter of certain hour constant to carry out root mean square calculation, obtain the average volume parameter of high and low frequency section audio data.

4, the method for automatic volume adjusting according to claim 1 is characterized in that, the control curve is an interactive knee adapt control curve in the step 3.

5, the method for automatic volume adjusting according to claim 1 is characterized in that, is transformed in the step 4 after the linear domain, carries out smoothing processing one time, multiplies each other again with through the corresponding frequency band data that postpone.

6, the method for automatic volume adjusting according to claim 5 is characterized in that, the method that described smoothing processing adopts is the low pass smothing filtering.

7, a kind of automatic volume regulating system receives the audio signal that A/D converter is exported, and carries out sending after automatic volume is regulated; It is characterized in that comprising the low pass filter, root mean square calculation module, control module and the multiplication module that connect successively, low pass filter blocks is connected to multiplier module by Postponement module; Also comprise the high pass filter, root mean square calculation module, control module and the multiplication module that connect successively, high pass filter block is connected to multiplier module by Postponement module; Described two multiplier modules are connected respectively to adder Module.

8, automatic volume regulating system according to claim 7 is characterized in that, described root mean square calculation module is the single order infinite impulse response filter with certain hour constant.

9, automatic volume regulating system according to claim 7 is characterized in that, described control module comprises the log-domain modular converter that connects successively, dynamic control law and threshold module and linear domain modular converter; Dynamically design interactive knee adapt control curve in control law and the threshold module makes the log-domain voice data that receives take parameter on trendline according to interactive knee adapt control, obtains ride gain.

10, automatic volume regulating system according to claim 7 is characterized in that, between described control module and the multiplier module, also has the smoothing filter module, on linear domain voice data is carried out smoothing processing.

11, automatic volume regulating system according to claim 10 is characterized in that, described smoothing filter module is to have the single order wireless pulses respective filter of different time constant for the signal increase and the situation that reduces.

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