KR0176640B1 - Audio signal encoding circuit - Google Patents

Abstract

[Technical field to which the invention described in the claims belong]

Audio signal coding circuit.

[Technical Challenges to Invent]

The use of masking phenomena and audible limits found in psychoacoustics to increase compression rates can satisfy the average listener's ears, but cannot hear sounds lower than the audible limits or masking curves. This can be a complaint to advanced users with sensitive hearing.

[Summary of the solution of the invention]

An audio signal encoding circuit, which encodes an input digital audio signal, decodes a first encoder that causes loss of information through the encoding, and outputs the output of the first encoder, wherein the loss of information occurs through the decoding. A first decoder, a subtractor for obtaining a difference between the audio signal output from the first decoder and the input digital audio signal, a second encoder for encoding the subtraction result without loss of information, and an output from the first encoder And means for selectively outputting the audio signal and the audio signal output from the second encoder according to a predetermined format.

[Important Uses of the Invention]

It is used for products requiring high efficiency audio signal encoding such as business recording / playback equipment or high quality audio equipment for high sound quality.

Description

Audio signal encoding circuit

1 is a block diagram of an encoder according to the present invention.

2 is a block diagram of a decoder according to the present invention.

* Explanation of symbols for main parts of the drawings

10: first encoder 20: first decoder

30: second encoder 40: subtractor

70: second decoder 80: adder

90: third decoder X: original signal

Xcom: Compression Signal

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a circuit for encoding an audio signal, and more particularly, to an audio signal encoding circuit which is appropriately adjusted to each other as necessary to have high and low information loss and compression ratios.

Conventional coding methods using acoustic psychology compress signals by removing and encoding fine signals that are determined to be inaudible to human hearing limits. That is, compression is performed by using an audible sound generated by the interference between the audible threshold volume and each frequency component. Although ATRAC, PASC, MUSICAM, and AC-3 using this method have already been recognized as having no problem at the general sound quality level, some users of advanced sound equipment do not tolerate these audio coding methods. Only raw-PCM claims to be able to produce the original sound. Therefore, an encoding method has been developed to increase the compression ratio without any loss of information so that these consumers can accept it, and this coding method can be used for the development of high-quality media with high sampling frequency and number of bits even with limited media capacity.

Meanwhile, in the conventional acoustic coding method, an encoder outputs a compressed signal through a process such as frequency domain transformation, masking processing, bit allocation, formatting / framing, multiplexing, and the like, and a decoder. ) Outputs an audio signal in the form of raw-PCM through demultiplexing, deframing, requantization, and time domain conversion. In order to increase the compression rate, masking phenomena and audible limits, which are revealed in acoustic psychology, can remove a large number of listeners, which can satisfy a large number of listeners, but they cannot hear sound at lower levels than the audible limit or masking curve. Is a frustrating affair for advanced users with sensitive hearings.

Accordingly, an object of the present invention is to secure a sound quality such as low-PCM by preserving through the appropriate compression suitable for the signal without discarding the fine signals that are considered inaudible in the conventional sound coding, and low access to conventional coding The present invention provides a high efficiency audio signal encoding circuit capable of transmitting and storing audio information at a bit rate.

According to an aspect of the present invention, an audio signal encoding circuit encodes an input digital audio signal, decodes a first encoder and loss of information through the encoding, and decodes an output of the first encoder. A first decoder having loss of information through the decoding, a subtractor for obtaining a difference between the audio signal output from the first decoder and the input digital audio signal, and a second encoder encoding the subtraction result without loss of information And an selecting means for selectively outputting the encoder and the audio signal output from the first encoder and the audio signal output from the second encoder according to a predetermined format.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings. In the following description, numerous specific details such as specific components of the circuit, addresses, and the like appear, which are provided to help a more general understanding of the present invention, and the present invention may be practiced without these specific details. It will be obvious to those of ordinary skill in the field. In the following description of the present invention, if it is determined that a detailed description of a related known function or configuration may unnecessarily obscure the subject matter of the present invention, the detailed description thereof will be omitted.

1 is a block diagram of an encoder according to the present invention.

If the transfer function of the first encoder 10 having a predetermined information loss is f (x), the transfer function of the first decoder 20, which also has a predetermined information loss due to the information loss, is f-1 (x). Since it cannot be, let h (x). On the other hand, if the transfer function of the second encoder 30 without information loss is set to g (x), the transfer function of the second decoder 70 without information loss becomes g-1 (x). If the input signal of the encoding circuit shown in FIG. 1 is x, the signal passed through the first encoder 10 becomes f (x), and the signal passed through the first decoder 20 is h (f (x)). Becomes The input of the lossless second encoder 30 through the subtractor 40 is xh (f (x)) and the output of the second encoder 30 is g (xh (f (x)). That is, one input of the multiplexer 50 is f (x) and the other input is g (xh (f (x)). The multiplexer 50 selectively outputs one of the two inputs according to a format (Xcom). do.

2 is a block diagram of a decoder according to the present invention. When the encoder output Xcom of FIG. 1 passes through the separating means, i.e., the demultiplexer 60, again the signal f (x) equal to the output of the first encoder 10 and g equal to the output of the second encoder 30 xh (f (x)) is separated The first encoding signal f (x) becomes h (f (x)) through the first decoder 20 having a predetermined information loss, and g (xh (f) below. (x)) becomes xh (f (x)) through g-1 (x), which is a transfer function of the second decoder 70 without loss of information, in the adder 80, two inputs h (f (x)). ) And xh (f (x)) are added to finally output the same signal x as the input of the above-described encoding circuit.

The invention can be used in several sectors, such as audio compression, stretching, and the like. It is especially used for digital audio products for high sound quality, and as the efficiency increases, the number of beats and the sampling frequency can be increased to record more excellent music signals on the same medium. This feature can be used in business recording / playback equipment or high-end audio equipment for high sound quality.

As described above, the present invention has an advantage of high efficiency as necessary by appropriately combining encoding / decoding with or without loss of information in encoding a digital audio signal.

Meanwhile, in the detailed description of the present invention, specific embodiments have been described, but various modifications are possible without departing from the scope of the present invention. Therefore, the scope of the present invention should not be limited to the described embodiments, but should be defined not only by the scope of the following claims, but also by the equivalents of the claims.

Claims (2)

An audio signal encoding circuit, which encodes an input digital audio signal, decodes a first encoder that causes loss of information through the encoding, and outputs the output of the first encoder, wherein the loss of information occurs through the decoding. A first decoder, a subtractor for obtaining a difference between the audio signal output from the first decoder and the input digital audio signal, a second encoder for encoding the subtraction result without loss of information, and an output from the first encoder And selection means for selectively outputting the audio signal and the audio signal output from the second encoder according to a predetermined format. The method of claim 1, wherein the third decoder is configured to decode the audio signal output from the first encoder and the information is lost through the decoding, and the audio signal output from the second encoder is decoded without loss of information. A fourth decoder for transmitting the audio signal output from the first encoder or the second encoder transmitted through the selection means to the third and fourth decoders, respectively, and the third and fourth decoders. And an adder for adding the audio signal output from the 4 decoder to generate the same digital audio signal as the input of the first encoder.