KR930006749B1 - Coding system for transmitted of digital audio - Google Patents

Abstract

The system for transmitting the TV digital audio signal without step-size information in the low data bit rate includes: a whole bandwidth amplification compressor (7) to compress tho audio noise rejected analog audio input by its signal level; a signal delay unit (8) to delay the compressor (7) outputs for the pre- determined time interval; a linear delta modulator for producing a digital audio bit stream by modulating the time-delayed input signals.

Description

Digital audio transmission encoder system

1 is a system configuration diagram of a conventional digital audio transmission encoder.

2 is a characteristic curve diagram of the first degree variable pre-emphasis means.

3 is a system configuration diagram of a digital audio transmission encoder of the present invention.

4 is a detailed configuration of the third full-band amplitude compression means.

5 is a detailed block diagram of the linear delta modulation means of FIG.

6 is a characteristic curve diagram of the full width amplitude compression means of FIG.

* Explanation of symbols for main parts of the drawings

1: low pass filter means 2: first signal delay means

3: variable pre-emphasis means 4: spectrum analysis means

5: bandwidth limiting means 6: analogue / digital conversion means

7: full-band amplitude compression means 8: second signal delay means

9: linear delta modulation means

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to the encoding of digital audio signals. In particular, the present invention relates to digital audio for transmitting an encoded digital audio signal by maximally reducing the information bit rate without the need for data for step size information transmission. A transmission encoder system.

As shown in FIG. 1, the conventional digital audio transmission encoder system includes a low pass filter means 100 for low-pass filtering an analog audio signal input from the outside to remove unnecessary signal components, and the low pass filter means ( And analyzing and extracting sound source information from the first signal delay means 101 for delaying and outputting the analog audio signal filtered and outputted from 100) for a predetermined time and the analog audio signal through the low pass filter means 100. In the spectrum analysis means 103, the first bandwidth limiting means 107 for removing sound source information below a predetermined level from the sound source information analyzed by the spectrum analysis means 103, and in the spectrum analysis means 103 Analyze the analog audio signal output from the first signal delay means 101 according to the extracted sound source information and remove the noise component through the emphasizing process Is a variable pre-emphasis means 102, second signal delay means 104 for delaying and outputting an analog audio signal output from the variable pre-emphasis means 102 for a predetermined time, and the variable pre-emphasis means. A step size determining means 105 for determining a step size of the analog audio signal outputted at 104 and an analog step size signal determined at the step size determining means 105 are converted into digital information. An analog / digital converting means 106 for outputting, second bandwidth limiting means 108 for removing a signal below a predetermined level among the step size signals determined by the step size determining means 105, and the second bandwidth limiting means ( Adaptive delta for delta-modulating the analog audio signal output from the second signal delaying means 104 and outputting it as a digital audio signal according to the step size signal output from 108). Second analog / digital converting means 110 for converting the analog sound source signal extracted by the tuning means 109 and the spectrum analysis means 103 into a digital audio signal in synchronization with a clock signal CLK input from the outside; And digital delay means 111 for delaying the digital signal output from the second analog / digital conversion means 110 for a predetermined time and outputting spectrum information.

In the drawing, reference numeral 200 denotes a preprocessing means including a first signal delay means 101, a spectrum analysis means 103, a first bandwidth limiting means 107, and a variable pre-emphasis means 102.

As described above, a system of the conventional digital audio transmission coder constructed as follows will be described with reference to FIG.

First, when an analog audio signal is input from the outside, the low pass filtering means 100 low-pass filters the input analog audio signal to an audible frequency level to remove unnecessary signal components, and then the first signal delay means 101 and Input to the spectrum analysis means 103.

The first signal delay means (101) delays the input analog audio signal by a predetermined time and inputs it to the variable pre-emphasis means (102), and the spectrum analysis means (103) from the low pass filter means (100). The actual sound source information signal is extracted from the input analog audio signal and input to the first bandwidth limiting means 107 and input to the second analog / digital converting means 110.

The first bandwidth limiting means 107 removes a sound source information signal having a predetermined level or less from the sound source information extracted by the spectrum analysis means 103 and only the sound source information signal having a predetermined level or more to the variable pre-emphasis means 102. Will be entered.

Accordingly, the variable pre-emphasis means 102 has a characteristic change according to the sound source information input from the first bandwidth limiting means 107 and thus the signal-to-noise ratio of the analog audio signal input from the first signal delay means 101. Will improve.

That is, the signal-to-noise ratio of the analog audio signal input from the first signal delaying means 101 becomes higher as the high frequency component increases, so that the variable pre-emphasis means 102 receives a constant input from the first bandwidth limiting means 107. As shown in FIG. 2, the characteristic is changed toward the negative decibel with respect to the "0" decibel according to the sound source information signal above the level, and the signal-to-noise ratio is improved. By varying the characteristic toward the positive decibel, the signal-to-noise ratio due to the noise masking effect, i.e., the audible noise is reduced and input to the second signal delay means 104 and the step size determining means 105.

The step size determining means 105 comprises an inclination detector for detecting the time derivative or inclination of the audio input signal from the variable pre-emphasis means 102.

The gradient detector generates a control signal indicating the step size to be used in the adaptive delta modulator 109 at this rear end.

The control signal output from the step size determining means 105 is limited by the second bandwidth limiting means 108 and input to the adaptive delta modulating means 109.

The first analog / digital converting means 106 converts the step size control signal coming from the step size determining means 105 into a digital signal of a bit stream to transmit the step size information to a decoder in the listener system. Done.

Meanwhile, the audio input signal delayed by the second signal delay means 104 is converted into a digital audio signal of the bit stream by the adaptive delta modulation means 109 according to the step size determined by the second bandwidth limiting means 108. And transmitted to a decoder in the listener system.

The reason for using the adaptive delta modulator 109 is to prevent the overslope phenomenon (more than the inclination reference value) when the slope of the analog audio input signal coming from the second signal delay means 104 is large. This requires adjustment of the step size.

In addition, the second analog / digital converting means 110 converts the spectrum of the sound source information extracted by the spectrum analyzing means 103 into a digital signal according to an external clock signal CLK and converts it into a digital delay means 111. Will be entered.

The digital delay means 111 transmits the spectral information of the second analog / digital conversion means 110 in time delay according to the input external clock signal CLK.

However, such a conventional digital audio transmission coder system transmits an audio bit stream using an adaptive delta modulation means, thereby bringing the complexity of hardware for controlling the step size and transmitting this information. Not only will this cause an increase, but if more channels for digital audio transmission are added to the same frequency spectrum as NTSC broadcast signals, bit errors will occur significantly.

Accordingly, an object of the present invention is to provide a digital audio transmission encoder system for transmitting a digital audio signal to be encoded to a decoder in a viewer system by reducing the information bit rate to the maximum without requiring data for transmitting step size information during transmission of an audio signal of a TV. have.

Means for achieving the object of the present invention is a full-band amplitude compression means for receiving an analog audio signal from which the audible noise has been removed by the signal processing from the pre-processing means and compressing the amplitude according to the level of the signal, the full-band A signal delay means for delaying output of an audio signal output from an amplitude compression means for a predetermined time and a linear delta modulation means for linearly modulating the time-delayed audio signal and outputting the digital audio bit stream are provided. When described in detail based on the accompanying drawings as follows.

3 is a system configuration diagram of a digital audio transmission encoder according to the present invention. As shown therein, a low pass filter means 1 for low pass filtering an input analog audio signal to remove unnecessary signal components, and the low pass filter means 1 Preprocessing means 20 for outputting by reducing the audible noise from the analog audio signal is filtered from the output and extracting the sound source information, and according to the level of the audio signal is reduced audible noise from the preprocessing means 20 A full bandwidth amplitude compression means (7) for compressing the amplitude and reducing the tilt angle; a second signal delay means (8) for delaying and outputting the audio signal output from the full bandwidth amplitude compression means (7) for a predetermined time; A linear delta edge for linearly modulating the time-delayed audio signal in the signal delay means 8 and outputting it as a digital audio bit stream to a decoder in the listener system. Means 9 and the analog / digital converting means 6 and the analog for converting the analog sound source signal extracted by the preprocessing means 20 into a digital audio signal in synchronization with a clock signal CLK input from the outside. And digital delay means (10) for delaying the digital signal output from the digital conversion means (6) for a predetermined time and outputting the spectrum information to the decoder in the listener system.

The preprocessing means 20 is a first signal delay means (2) for delaying and outputting the analog audio signal filtered out from the low-pass filter means (1) for a set time, and the low-pass filter means (1) Spectrum analysis means (4) for analyzing and extracting sound source information from the analog audio signal through, and bandwidth limiting means (5) for removing sound source information below a predetermined level from the sound source information analyzed and extracted by the spectrum analysis means (4) And variable pre-emphasis means (3) for outputting the analog audio signal output from the first signal delay means (2) by removing the noise component through an emphasis process according to the sound source information limited by the bandwidth limiting means (3). ).

In the above-described full-band amplitude compression means 7, the audible noise is reduced by the variable pre-emphasis means 3, as shown in FIG. 4, according to the peak value fed back to the audio signal. A voltage filter amplifier 7a for controlling gain and amplifying and outputting the second signal delay means 8; and a band filter for band-filtering the output signal of the voltage control amplifier 7a to remove a signal below a predetermined level. 7b, a rectifier 7c for converting the output signal of the band pass filter 7b to a direct current level, and a peak detector 7d for detecting peak values of the converted direct current level and inputting them to the voltage control amplifier 7a. ).

The linear delta modulating means (9) comprises a low pass filter (9a) for low-passing the audio signal delayed in the second signal delay means (8), as shown in FIG. A differential amplifier 9b for amplifying and outputting the output audio signal of 9a) from the reproduction integral value with respect to the audio signal, and comparing the difference signal of the differential amplifier 9b with the input audio signal and the resulting signal. A comparator 9c for outputting a digital signal, a flip-flop 9d for converting an analog audio signal output from the comparator 9c into digital audio, and outputting A multiplier 9e for multiplying the size control signal and an integrator 9f for integrating the output signal of the multiplier 9e and inputting them to the differential amplifier 9b.

Thus, the operation and effect of the present invention configured in detail as follows.

First, when an analog audio signal is input from the outside, the low pass filtering means 1 low-pass filters the input analog audio signal to an audible frequency level to remove unnecessary signal components, and then the first signal of the preprocessing means 20. It is input to the delay means 2 and the spectrum analysis means 4.

The first signal delay means (2) delays the input analog audio signal by a predetermined time and inputs it to the variable pre-emphasis means (3), and the spectrum analysis means (4) from the low pass filter means (1). The actual sound source information signal is extracted from the input analog audio signal and input to the bandwidth limiting means 5, and input to the analog / digital converting means 6 as well.

The bandwidth limiting means 5 removes a sound source information signal of a predetermined level or less from the sound source information extracted by the spectrum analysis means 4 and inputs only the sound source information signal of a predetermined level or more to the variable pre-emphasis means 3. do.

Accordingly, the variable pre-emphasis means (3) is changed in accordance with the sound source information input from the bandwidth limiting means (5) to improve the signal-to-noise ratio of the analog audio signal input from the first signal delay means (2) Done.

In detail, the variable pre-emphasis means (3) is inputted from the bandwidth limiting means (5) because the signal-to-noise ratio worsens as the high frequency component of the analog audio signal input from the first signal delay means (2) increases. As shown in FIG. 2, the characteristic is changed to 88 to 96 minus centering on the "0" decibels according to the sound source information signal of a predetermined level or more, and the signal-to-noise ratio is improved. The characteristic is changed to 84 or 86 of positive about the decibels so that the signal-to-noise ratio due to the noise masking effect is reduced, i.e., the audible noise is reduced and input to the voltage control amplifier 7a of the full-band amplitude compression means 7 of FIG. .

As shown in FIG. 6, the characteristic of the full-band amplitude compression means 7 is to adjust the level of the signal only according to the signal level regardless of the frequency of the input audio signal. The smaller the signal, the smaller the level, the larger the level.

Referring to FIG. 4, the input audio signal from the variable pre-emphasis means (3) is input to the voltage control amplifier (7a) of the full-band amplitude compression means (7). 7a) adjusts and outputs the gain according to the feedback input of the peak value with respect to the input audio signal.

That is, the compressed audio signal output by controlling the gain from the voltage control amplifier 7a is filtered through the band filter 7b so that only a signal of a predetermined level or more is converted into a DC level through the rectifier 7c, and thus the peak detector. It is input to 7d.

The peak detector 7d detects the peak value of the DC level input from the rectifier 7c and inputs it to the voltage control amplifier 7a.

The voltage control amplifier 7a compresses the input audio signal as shown in FIG. 6 according to the peak value input from the peak detector 7d.

That is, if the control voltage from the peak detector 7d is inputted high (H), the gain is decreased and output. If the control voltage is low (L), the gain is increased and the amplitude of the input audio signal is compressed and output. .

The transfer characteristic of the full-band amplitude compression means 7 as described above is as follows.

Since the transmission characteristic is A (a) = 1 / a (a = 10 ^0/20 ): c is the decibel-ms value of the signal amplitude, the linear delta modulation means ( 9), only the fixed step size is properly selected so as not to cause overslop.

The appropriately compressed input audio signal in the full-band amplitude compression means 7 is time-delayed through the second signal delay means 8 and filtered in the low pass filter 9a of the linear delta modulation means 9 as shown in FIG. Only the audible frequency signal is input to the differential amplifier 9b.

The differential amplifier 9b amplifies the difference between the audio signal input from the low pass filter 9a and the reproduction integral value for the audio signal and inputs it to the comparator 9c.

The comparator 9c compares the difference signal of the differential amplifier 9b with the input audio signal and outputs "0" if the input audio signal is larger than the reproduction signal, i.e., the difference signal, and outputs "0" if it is smaller. Input to flop 9d.

The flip-flop 9d transmits the input analog audio signal as a digital audio bit stream to a decoder inside the listener system and inputs it to the multiplier 9e.

The multiplier 9e multiplies the input digital audio signal with an appropriately fixed step size control signal from the outside and inputs it to the integrator 9f.

The integrator 9f reproduces the signal input from the multiplier 9e as an original audio signal and inputs it to the differential amplifier 9b, so that the above operation is repeated and the digital audio bit stream is transmitted.

Therefore, in the decoder in the chord extraction system, by constructing an expander that works opposite the compression means, the original signal can be reproduced.

The encoder of the present invention is a part of a broadcasting station for transmitting audio bits and spectral information bits to a decoder of a listener system.

As described in detail above, the present invention does not require data for step size information transmission, thereby reducing information bit rate and eliminating the need for a separate hardware for step size control. There is an effect that can suppress the increase of the channel for.

Claims (3)

Full bandwidth for receiving an analog audio signal that is processed through signal delay means, spectrum analysis means, bandwidth limiting means, and variable pre-emphasis means included in the preprocessing means, and audible noise is removed, and compressing the amplitude according to the level of the signal. An amplitude compression means, a signal delay means for delaying the audio signal output from the full bandwidth amplitude compression means for a predetermined time, and a linear delta modulation means for linearly modulating the time-delayed audio signal and outputting it as a digital audio bit stream; Digital audio transmission encoder system. 2. The voltage control method of claim 1, wherein the full-band amplitude compression means adjusts the gain according to the peak value fed back to the audio signal by reducing the audible noise in the preprocessing means and amplifying the signal to the signal delay means. An amplifier, a band pass filter for band-filtering the output signal of the voltage control amplifier to remove a signal below a predetermined level, a rectifier for converting the output signal of the band pass filter to a DC level, and a peak value of the converted DC level. And a peak detector for detecting and inputting the signal to a voltage controlled amplifier. 2. The apparatus of claim 1, wherein the linear delta modulator comprises: a low pass filter for low pass filtering the input audio signal delayed by the signal delay means; and a difference between the output audio signal of the low pass filter and the reproduction integral value for the audio signal. A differential amplifier for amplifying and outputting a comparator for comparing the difference signal of the differential amplifier and an input audio signal and outputting a result signal, and a flip-flop for converting and outputting an analog audio signal output from the comparator to digital audio And a multiplier for multiplying the flip-flop digital audio signal and an external step size control signal, and an integrator for reproducing the output signal of the multiplier as an original audio signal and inputting it to a differential amplifier. Audio transmission encoder system.

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