KR19990016522A - Analog 3-D Sound System Using TLA - Google Patents

Abstract

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a three dimensional sound system, and more particularly to a three dimensional sound system using a Table Lookup Algorithm (TLA) capable of processing analog inputs. Are converted through the first and second ADCs 71 and 72, respectively, and addressed to the parameter ROM 74. In response, coefficient data of the parameter ROM 74 is output. These are respectively input to the first and second DLPFs 77 and 78, filtered and decoded, and input to the first and second PGMMIXs 82 and 84, respectively. The first and second PGMMIXs 82 and 84 multiply them with the left and right input signals Lin and Rin and add the results to output the left and right three-dimensional voice signals Lout and Rout.

Description

Analog 3-D Sound System Using TLA

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a three dimensional sound system, and more particularly, to a three dimensional sound system using a Table Lookup Algorithm (TLA) capable of processing an analog input.

The three-dimensional sound system is a system for reproducing a sound closer to a sound source based on a human hearing system. This is a method for reproducing a higher sound quality with only two speakers, such as SRS, Qsound, and 3D spatializer Table Lookup Architecture (TLA).

In particular, unlike the conventional TLA, a left / right stereo signal is converted into three-dimensional sound by using a separate lookup table stored in a ROM.

1 is a circuit diagram of a conventional digital TLA three-dimensional sound system.

As shown in FIG. 1, a conventional digital TLA three-dimensional sound system includes first and second log scale converters 10 and 20 and a parameter ROM 30. And the first and second digital low pass filters (DLPFs) 40 and 50, the first to fourth multipliers 60, 62, 64 and 66, and the first and second adders 70 and 72. Include.

In the three-dimensional sound system using the TLA as described above, the left and right voice input signals R (i) and L (i) are multiplied by coefficients (α, β) determined by the magnitude of each input signal, and the results are added separately. To print. In this case, the final output result is shown in Equation 1 below.

Lout = a * Lin + b * Rin

Rout = b * Lin + a * Rin

In Equation 1, coefficients a and b are coefficients for converting to a sound source closer to a sound source based on Psycho Acoustic. These values are stored in the parameter ROM 30. In general, it is stored in the parameter ROM (30). In general, the parameter ROM 30 is addressed as 8 * 8 and divided into 8 levels according to the magnitudes of the left and right signals to multiply coefficients.

The conventional TLA method as described above receives a digital signal, converts it into a three-dimensional sound, and outputs the digital signal. In addition, since all voice signals in the natural world are analog signals, a digital to analog converter (DAC) for converting a digital signal into an analog signal is provided to realize an actual voice. Input from the play path (DAC path) of a sound card mounted on a personal computer or the like is a variety of analog signals are input to the analog mixer (analog mixer) and gain / attenuation adjustment The final output is via gain / attenuation.

However, when the conventional TLA method is used, a problem occurs in processing of 3D audio) by converting the sound card into 3D audio only before the DAC input stage only for pulse code modulation (PCM) input of the sound card. Therefore, the three-dimensional voice signal had to be implemented in analog.

Accordingly, an object of the present invention is to provide an analog three-dimensional sound system using a TLA that can be processed in an analog manner as proposed to solve the above-mentioned problems.

1 is a circuit diagram of a conventional digital TLA three-dimensional sound system,

2 is a circuit diagram of an analog three-dimensional sound system according to a preferred embodiment of the present invention;

3 is a detailed circuit diagram of an example of implementing the PGAMIX shown in FIG.

Explanation of symbols for the main parts of the drawings

10, 20: log scale converter 30, 74: parameter ROM

40, 50, 77, 78: DLPF 60, 62, 64, 66: Multiplier

70, 72: Adder 82, 84: PGAMIX

According to a feature of the present invention as described above, a three-dimensional sound system using a TLA includes: an analog-to-digital converter for receiving a left and right voice signal; A parameter ROM addressed according to the output of the ADC to output corresponding coefficient data; A digital low pass filter for filtering coefficient data output of the parameter ROM, decoding the predetermined number of bits, and outputting the predetermined number of bits; And a programmable enable attenuator and mixer for multiplying the output of the digital low pass filter and the left and right voice signals and adding the results to the left and right 3D sound signals.

In this embodiment, the ADC includes a first ADC for receiving a left voice signal; And a second ADC receiving a right voice signal.

In this embodiment, the ADC receives the left (right) voice signal during each allocated time by using a time allocation method and converts the analog-digital signal.

According to the present invention as described above, the left and right input signals Lin and Rin are converted through the first and second ADCs, respectively, and addressed to the parameter ROM. In response, coefficient data of the parameter ROM is output. These are respectively input to the first and second DLPFs, filtered and decoded, and input to the first and second PGMMIXs, respectively. The first and second PGMMIX multiply this by the left and right input signals Lin and Rin and add the result to output the left and right three-dimensional voice signals Lout and Rout.

(Example)

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

FIG. 2 is a circuit diagram of an analog three-dimensional sound system according to a preferred embodiment of the present invention, and FIG. 3 is a detailed circuit diagram of an example of implementing the PGMMIX shown in FIG.

As shown in FIG. 2, the analog three-dimensional sound system using the novel TLA of the present invention includes the first and second ADCs 71 and 27, the parameter ROM 74, and the first and second DLPFs 77,. 78), and PGMMIX (Programmable Gain Attenuator and Mixer) 82, 84.

Referring to FIG. 2, the parameter ROM 74 and the first and second DLPFs 77 and 78 have the same configuration as the conventional configuration shown in FIG. The first and second DLPFs 77 and 78 prevent a sudden change in coefficient change. That is, when the change of coefficient is abrupt, the phenomenon which appears as a noise in actual voice signal processing is suppressed. Therefore, in order to obtain the effect of integrating the coefficient, it can be configured as the first-order Infinite Impulse Response type.

First, left and right voice signals Lin and Rin are input to the first and second ADCs 71 and 72, respectively. The first and second ADCs 71 and 27 may be configured as ADCs having an 8-bit resolution, and their outputs are input to the parameter ROM 74. Since the coefficients stored in the parameter ROM 74 are encoded at a log scale of an input, the first and second inverters 71 and 72 are configured in a logarithmic type to suit this. . In addition, since the band of the input signal is 20KHz, the sampling frequency is about 100KHz. In addition, since it operates in a low frequency band and has a resolution of 8 bits, each ADC of the system can be implemented with one ADC using time sharing.

Subsequently, a corresponding address area of the parameter ROM 74 is designated according to the characteristics of the input signals Lin and Rin to select data. This is filtered through the first and second DLPFs 77 and 78, decoded into 5 bits, and input to the first and second PGMMIX 82 and 84, respectively. Each input signal is multiplied by the left and right input signals Lin and Rin, and the result is added to output the Lout and Rout signals.

3 is a circuit diagram of an example of the PGMMIX.

Referring to FIG. 3, when implementing a PGA suitable for each bit number, the resistor and transistor array may be configured to be suitable for the bit number. For example, if you want to implement a 4-bit PGA, you can use 16 resistor and transistor arrays. If you want to use 5 bits, expand to 32. The product of each coefficient is summed using an operational amplifier and outputs Lout and Rout signals.

And when the combination of coefficients to be transformed changes, simply change the parameter ROM. When the number of bits of the coefficient data of the parameter ROM increases, the number of resistors of the PGA may be extended by an index of two. For example, for 6 bits, 64 resistors can be used.

According to the present invention as described above, the three-dimensional sound system using the TLA of the present invention, which can be used for a sound card, etc., enables three-dimensional sound with respect to an analog input.

Claims (3)

An analog-to-digital converter that receives left and right voice signals Lin and Rin; A parameter ROM addressed according to the output of the ADC to output corresponding coefficient data; A digital low pass filter for filtering coefficient data output of the parameter ROM, decoding the predetermined number of bits, and outputting the predetermined number of bits; A programmable gain attenuator and mixer for multiplying the output of the digital low pass filter and the left and right voice signals Lin and Rin and adding the result to the left and right 3D sound signals Lout and Rout. 3D sound system using TLA. 2. The ADC of claim 1, further comprising: a first ADC (71) for receiving a left voice signal; 3D sound system using a TLA comprising a second ADC (71) receiving a right voice signal. The 3D sound system of claim 1, wherein the ADC receives a left-right voice signal during each allocated time by using a time allocation scheme and performs analog-to-digital conversion.