KR100195712B1 - Acoustoptic control apparatus of digital audio decoder - Google Patents

Abstract

The present invention relates to a sound quality control apparatus of a digital audio decoder, which includes a bit allocation decoder (10), a dequantizer (12), a scale factor decoder (14), a sound quality control unit (15), and a synthesis subband filter ( 16).

The dequantizer dequantizes each subband sample of the encoded bit stream according to the bit allocation information, and the scale panter decoder 14 uses the scale factor to originalize each subband sampled in the dequantizer 12 using the scale factor. The sound quality control unit 15 obtains a predetermined weight according to the sound quality control signal input from the outside, and then adds the weight to the coefficient of each subband sample output from the scale factor decoder 14. Adjust the sound quality by multiplying. The synthesized subband filter 16 synthesizes each subband sample output from the sound quality control unit 15 to restore an audio signal.

Therefore, according to the present invention, the sound quality can be easily adjusted by a digital operation of multiplying a predetermined weight according to the sound quality control signal in the digital audio decoder.

Description

Sound quality control device of digital audio decoder

1 is a block diagram of a typical MPEG audio encoder.

2 is a block diagram of a typical MPEG audio encoder.

3 is a block diagram of an MPEG audio decoder equipped with a sound quality control apparatus according to the present invention.

4 shows frequency characteristics of a weight function according to the present invention.

(a) shows the frequency characteristic of the weight function at the time of bass enhancement,

(b) shows the frequency characteristics of the weight function during bass attenuation,

(c) shows the frequency characteristics of the weight function in treble enhancement,

Figure (d) shows the frequency characteristics of the weight function at high attenuation.

* Explanation of symbols for main parts of the drawings

2: filterbank 4: psychoacoustic model

6 bit allocation and quantizer 8 bit stream format

10: bit allocation decoder 12: dequantizer

14: scale factor decoder 15: sound quality control unit

16: synthetic subband filter

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital audio decoder, and more particularly, to an apparatus for controlling sound quality of a digital audio decoder capable of simply adjusting sound quality by digital operation in a digital audio decoder.

Digital audio became the standard for audio equipment in the 80s with the development of mass storage media such as compact discs (CDs) and digital audio tapes (DATs). However, since digital audio data has a large amount of information, it is essential to compress audio data for use in a medium having a limited bandwidth such as over-the-air broadcasting.

Therefore, various high quality audio compression technologies have been developed since the late 80's, and these technologies commonly take human hearing characteristics into existing data compression techniques.

A widely known audio compression technology is the MPEG standard defined by the MPEG / ISO which determines the standard of the video and the audio compression method added thereto. In other words, MPEG-1 is a coding scheme capable of compressing video and audio at about 1.5 Mbit / s, and uses MUSICAM (Masking-pattern adapted Univesal Subband Integrated Coding And Mulitplexing). This has been extended to multi-channel MPEG-2 with a data rate of over 6 Mdit / s for digital broadcasting.

The MUSICAM method is a subband coding method using auditory characteristics, and it is possible to obtain a reconstruction sound that is subjectively identical to the original sound at 96 to 128 Kbit / s.

As shown in FIG. 1, the MPEG audio coder using the MUSICAM scheme includes: a filter bank 2 for converting an input signal (PCM data) into a plurality of subband samples and outputting the same; Fast Fourier Transform (FFT) of the input signal to obtain spectral information, obtain a masking threshold from the spectral information, and then obtain the difference between the masking threshold and the sound pressure level of each subband sample determined from the spectral information. A psychoacoustic model 4 for calculating and outputting a signal to mask ratio (SMR); Bit allocation and quantization for allocating bits to the respective subband samples using the signal-to-mask ratio (SMR) output from the psychoacoustic model 4 and quantizing and outputting the respective subband samples according to the allocated bits. Group 6; A bit stream format unit 8 for formatting and outputting additional information such as subband samples, bit allocation information, and scale information, which are quantized and output by the bit allocation and quantizer 6, into a bit stream. It is configured to include.

In the MPEG audio encoder configured as described above, the filterbank 2 stores 32 new audio samples which are sequentially input into a buffer having a size of 512 samples, multiplies the buffer by an analysis window, and then 512 The sample is divided into eight 64 sample blocks and each block is added to form a new vector.

This is multiplied by an analysis matrix such as the following equation 1 to make 32 subband samples.

The psychoacoustic model 4 then determines the maximum noise level that is masked by the original sound in each subband, and uses this noise level (masking threshold) to determine the actual number of quantized bits in each band. can do.

In the MPEG method, two psychoacoustic models are provided. The first psychoacoustic model distinguishes pure noise and noise components from a spectrum of a signal, calculates individual masking thresholds of pure noise and noise, and an absolute audible limit. In consideration of the above, the signal-mask ratio is obtained by calculating a total masking threshold and calculating a signal-to-mask ratio in each subband. That is, the spectrum is obtained through a fast Fourier transform (FFT), and the sound pressure level in each subband is determined therefrom.

In this case, since the masking curve varies depending on whether the masking component is pure sound or noise, the pure silver component and the noise component should be found from the spectral information. That is, if 7 dB or more of the partial signal (1ocal maxima) is larger than the surrounding signal, it is regarded as pure sound and finds the pure sound component, and then obtains one noise component within one critical band from the rest of the spectrum. Experimentally obtained masking function is applied to obtain the masking threshold by pure sound, and the total masking threshold is obtained by the sum of the individual masking threshold and the audible limit.

The signal to masking ratio (SMR) is calculated by calculating the difference between the sound pressure level in each subband and the masking threshold. As a result, if the signal-to-mask ratio (SMR) is small, the sound pressure level of the signal is small or masking is high, so that the effective quantization can be performed with fewer bits.

On the other hand, the second psychoacoustic model, the process of calculating the energy according to the critical band of the signal, the process of calculating the spreading function and convolution, the degree of excitation of the auditory nerve, and considering the absolute audible limit masking threshold value The signal-to-mask ratio is obtained through the calculation process and the process of calculating the signal-to-mask ratio in each subband.

This second psychoacoustic model convolves the fast Fourier transform spectrum with a spreading function, which is an excitation model of the auditory nerve, to obtain a masking threshold, thereby obtaining a large amount of computation but more accurate results.

Referring again to FIG. 1, the bit allocation and quantizer 6 uses each signal in the subband samples output from the filter bank 2 using the signal-to-mask ratio output from the psychoacoustic model 4. A bit is allocated for each subband sample and divided by a scale factor according to the allocated bit to quantize each normalized subband sample. At this time, the scale factor calculation for each subband is performed before quantization, and the scale factor calculation is performed every 12 samples, and the maximum value of the absolute values of 12 samples is found to be 0 to 2. Normalize to

In addition, the bit stream formatter 8 formats and transmits additional information such as quantized subband samples, bit allocation information, and scale information output from the bit allocation and quantizer 8 into a bit stream. .

Meanwhile, the digital audio decoder which restores the compressed and transmitted bit stream as described above may implement the digital audio encoder in reverse. That is, the conventional digital audio decoder includes a bit allocation decoder 10 for decoding and outputting bit allocation information for each subband sample in a compressed and transmitted bit stream as shown in FIG. 2; An inverse quantizer (12) for inversely quantizing each subband sample of the bit stream according to the bit allocation information output from the bit allocation decoder (10); A scale factor decoder (14) for detecting scale factor information in the bit stream and calculating and outputting each subband sample quantized by the inverse quantizer (12) using the scale factor as an original subband sample; Composed of a subband filter 16 for synthesizing each subband sample output from the scale factor decoder 14 to restore an audio signal.

In the MPEG audio decoder as described above, the bit allocation decoder 10 decodes and outputs bit allocation information for each subband sample among the compressed and transmitted bit streams, and the dequantizer 12 uses the bit allocation decoder ( The subband samples of the bit stream are inversely quantized and output according to the bit allocation information output from 10).

The scale factor decoder 14 detects scale factor information in the bit stream, and then calculates and outputs each subband sample dequantized by the dequantizer 12 as the original subband sample using the stale factor. The synthesized subband filter 16 synthesizes each subband sample output from the scale factor decoder 14 to restore and output an audio signal.

In this case, the matrix used for reconstructing each subband sample into an audio signal in the synthesis subband filter 16 is expressed by the following equation.

The PCM data decoded as described above is output to the speaker through a digital-analog converter and a low frequency amplifier circuit.

In this case, in the low frequency amplification circuit, a sound quality control circuit is provided to change the sound quality from the speaker according to a user's taste, and such sound quality control circuit applies a low sound attenuation, a high sound attenuation or a high sound augmentation circuit or a negative feedback (NFB). There is one, and the adjustment can be changed continuously, and there is a switch to switch to a predetermined circuit by the changeover switch.

However, the conventional sound quality control circuit as described above has a problem that the implementation of the circuit is complicated by controlling the sound quality by controlling the amplification of the analog-converted signal.

Accordingly, an object of the present invention is to provide a sound quality control apparatus of a digital audio decoder capable of simply adjusting sound quality by a digital operation in a digital audio decoder.

In order to achieve the above object, the present invention provides a digital audio decoder that receives a decoded digital audio bit stream, decodes bit allocation information, and decodes each subband sample of the bit stream encoded according to the bit allocation information. An inverse quantizer with inverse quantization; A scale factor decoder for calculating and applying a factor to each subband sample dequantized by the dequantizer; A sound quality control unit which obtains a predetermined weight according to a sound quality control signal input from the outside and adjusts sound quality by multiplying the weight of the coefficient of each subband sample output from the scale factor decoder; And a synthesized subband filter for reconstructing the audio signal by synthesizing each subband sample output from the sound quality control unit.

Hereinafter, with reference to the accompanying drawings will be described an embodiment of the present invention;

3 is a block diagram of an MPEG audio decoder equipped with a sound quality control apparatus according to the present invention, which includes a bit allocation decoder 10 for decoding and outputting bit allocation information for each subband sample from a compressed and transmitted bit stream; ; An inverse quantizer (12) for inversely quantizing each subband sample of the bit stream according to the bit allocation information output from the bit allocation decoder (10); A scale factor decoder 14 that detects scale factor information from the bit stream and then calculates and outputs each subband sample dequantized by the dequantizer 12 as an original subband sample using the steering factor. ; A sound quality control unit 15 for multiplying each subband sample output from the scale factor decoder 14 by a weight according to a sound quality control signal input from the outside; The synthesized subband filter 16 synthesizes each subband sample output from the sound quality control unit 15 to restore an audio signal.

At this time, the sound quality control unit 15 is configured to multiply the frequency coefficient of the subband sample by the weight according to the sound quality control signal input from the outside and output the weight coefficient. The sound quality control signal includes a low tone control signal for augmenting or attenuating a low tone.

Therefore, the sound quality control unit 15 multiplies the low frequency region coefficient of the subband sample by a predetermined high weight when a sound quality control signal (referred to as a 'low sound amplification signal') for augmenting low sound from the outside is received, and the sound quality for low sound attenuation. When a control signal (referred to as a "low attenuation signal") is input, the low frequency region coefficient of the subband sample is multiplied by a predetermined low weight to be output. When a sound quality control signal (referred to as a 'high sound amplification signal') for augmenting a high sound is input, the high frequency region coefficient of the subband sample is multiplied by a predetermined high weight, and a sound quality control signal for attenuating the high sound (this is referred to as a 'high sound attenuation signal'). When the signal is input, the high frequency region coefficient of the subband sample is multiplied by a predetermined low weight to be output.

The operation and effects of the sound quality control apparatus of the digital audio decoder according to the present invention will be described in detail as follows.

The bit allocation decoder 10 decodes and outputs bit allocation information for each subband sample from the compressed and transmitted bit stream, and the inverse quantizer 12 outputs the bit allocation information output from the bit allocation decoder 10. Accordingly, each subband sample of the bit stream is inversely quantized and output.

The scale factor decoder 14 detects scale factor information in the bit stream, and then calculates each subband sample dequantized by the dequantizer 12 as the original subband sample using the scale factor. Output

Meanwhile, the sound quality control unit 15 multiplies each subband sample output from the scale factor decoder 14 by a weight according to a sound quality control signal (bass or treble control signal) input from the outside to the synthesized subband filter 16. Output

In this case, the sound quality control unit 15 calculates a weight according to a sound quality control signal input from the outside, and then multiplies the weight by the frequency coefficients of the subband samples.

For example, assuming that the weights are from 0 to 2, and 1 is to output the input as it is, examples of weights according to the augmentation and attenuation of the bass and treble are shown in Table 1.

Weight table

Referring to Table 1, when the bass enhancement signal is input, the low frequency region coefficient of the subband sample is multiplied by a high weight (that is, 1.5). Multiply by 0.5) and output. In addition, when a high augmentation signal is input, the high frequency domain coefficient of the subband sample is multiplied by a high weight (that is, 1.5), and when a high attenuation signal is input, the high frequency domain coefficient of the subband sample is multiplied by a low weight (0.5). .

Such weight characteristics are as shown in (a) to (d) of FIG. Referring to (a) of FIG. 4, it can be seen that a relatively high weight is added to the low frequency region according to the low sound amplification signal. Referring to (b) of FIG. 4, it is relatively low to the low frequency region according to the low attenuation signal. It can be seen that the weighting. Referring to (c) of FIG. 4, it can be seen that a relatively high weight is added to the high frequency region according to the high sound amplification signal. Referring to (d) of FIG. 4, it is relatively high to the high frequency region according to the high attenuation signal. You can see that it gives a low weight.

Accordingly, the sound quality of the digital audio can be easily adjusted by obtaining weights according to the bass control signal and the treble control signal input from the outside, and then multiplying the weights by the frequency coefficients of the subband samples.

The synthesized subband filter 16 synthesizes each subband sample output from the sound quality control unit 15 to restore and output an audio signal.

In this case, the matrix used for reconstructing each subband sample into an audio signal in the synthesis subband filter 16 is expressed by the following equation.

As described above, according to the present invention, the sound quality can be easily adjusted by a digital operation of multiplying a predetermined weight according to the sound quality control signal in the digital audio decoder.

Claims (2)

A digital audio decoder that receives a compressed coded digital audio bit stream, decodes and then decodes bit allocation information, wherein the dequantizer 12 dequantizes each subband sample of the bit stream encoded according to the bit allocation information. Wow; A scale factor decoder 14 for calculating by applying a scale factor to each subband sample dequantized by the inverse quantizer 12; A sound quality control unit 15 which obtains a predetermined weight according to a sound quality control signal input from the outside and adjusts the sound quality by multiplying the weight of the subband sample output from the scale factor decoder 14 by the weight; And a synthesis subband filter (16) for reconstructing an audio signal by synthesizing each subband sample output from the sound quality control unit (15). The sound quality control signal of claim 1, wherein the sound quality control unit 15 multiplies the low frequency region coefficient of the subband sample by a predetermined high weight when a sound quality control signal for augmenting low sound is input from the outside, and attenuates the low sound. If is input, multiply the low frequency region coefficient of the subband sample by a predetermined low weight, and when the sound quality control signal for augmenting the high frequency is input, multiply the high frequency region coefficient of the subband sample by a predetermined high weight and attenuate the high sound. The sound quality control device of the digital audio decoder, characterized in that when the sound quality control signal for the input is multiplied by a predetermined low weight multi-frequency coefficient of the subband sample.