KR20040082905A - Device and Method for Digital Voice Amplification and Attenuation in Telephony Voice Network - Google Patents

Abstract

PURPOSE: An apparatus for amplifying and attenuating digital voice data in a telephone voice network, and a method thereof are provided to apply various applications in the case that voice amplification is digitally processed in an environment using a PLD(Programmable Logic Device) and VHDL(Very High speed IC Hardware Description Language). CONSTITUTION: A linear converter(202) converts an inputted signal(201) into a linear code. A sign/magnitude separator(203) separates a sign bit and a magnitude bit of the linear code. An amplifier and attenuator(204) shifts the magnitude bit to the right and left, and adds the shifted values for amplifying or attenuating the magnitude bit. A sign/magnitude combining unit(205) combines the sign bit with the amplified or attenuated magnitude bit for generating amplified or attenuated signal. A non-linear converter(206) executes inverse conversion for obtaining an amplified or attenuated signal(207).

Description

Device and method for digital voice data amplification / attenuation in telephone voice network {Device and Method for Digital Voice Amplification and Attenuation in Telephony Voice Network}

The present invention proposes a method for digitally amplifying and attenuating a voice of a caller transmitted digitally in a telephone network.

The method of amplifying and attenuating voice is generally an analog method using an analog amplifier. In the Modulation and Demodulation (MODEM), an automatic gain controller (AGC) provides this function. Analog amplification / attenuation methods add new noise, which not only damages the original sound but also requires many circuit elements to process multiple voice signals simultaneously. In the public network, digital data is already transmitted, and digital methods are applied to amplify and attenuate these voices. However, the necessity of digital voice amplification / attenuation is not required in the voice telephone network.

In case of digital processing of voice amplification in the environment using the latest PLD (Programmable Logic Device) technology and VHDL (Very high speed IC Hardware Description Language), it can be applied to various applications and overcomes the disadvantages of analog amplification / attenuation method. Not only can it be done, it is also advantageous in stable operation.

As the invention deals with digital signals directly, various applications are developed when digitally processing speech amplification in an environment using device development and experiments, PLD (Programmable Logic Device) technology, and VHDL (Very high speed IC Hardware Description Language). Not only can be applied to, but it can overcome the disadvantages of the analog amplification / attenuation method, and also process the digital signal directly, which is advantageous in device development, experiment and stable operation.

Figure 1: Functional block diagram of the device

2 is a block diagram of a digital voice amplification and attenuation method

Fig. 3: Example of Gaining Digital Voice Amplification and Attenuation

Figure 4: Code conversion table for converting linearly from A-Law and Mu-Law

Figure 5: Code conversion table for converting linear to A-Law and Mu-Law

In the case of voice, the Public Switched Telephony Network (PSTN) uses a non-linear A-Law or Mu-Law method of ITU-T Recommendation G.711 when converting analog signals to digital in consideration of ear characteristics and transmission bands. Use code Nonlinear codes are codes that are coded in a nonlinear form so that the actual loudness between codes is not equal. Therefore, in order to change the loudness of a voice, a nonlinear code must first be converted into a linear code, and then the converted linear code is changed. You must reassign the code to match the original nonlinear coding scheme.

1 is a functional block diagram of a device. In the figure, the processor unit 101 includes a commercially available processor such as Motorola's MPC8260, a flash ROM (Read Only Memory), and a RAM (Random Access Memory) to control the overall functions of the device through the processor bus 108. It is possible to store the machine voice data such as background music from the Ethernet in the RAM, and to transfer the machine voice data in the RAM to the switch unit through the serial bus, and received through the WAN (Wide Area Network) matching unit and the switch unit. Call processing data performs call processing related to call setup / release according to the call processing protocol. The WAN matching unit 102 matches the time division multiplex (TDM) based digital line signal and delivers serial voice data to the voice mixing unit. The voice mixing unit 103 performs voice mixing and voice data amplification / attenuation functions under the control of the processor unit. The voice mixing unit 103 receives the machine voice from the switch unit to perform amplification and attenuation functions, and the voice transmitted from the WAN matching unit. And transfer it to the switch unit or mix it with the voice transmitted from the switch unit to transfer to the WAN matching unit. The switch unit 104 performs a function of mutually inputting a digital signal of a processor unit, a voice mixer and a signal processor through a serial bus. The signal processing unit 105 detects or transmits a dual tone multi-frequency (DTMF) signal from digital data input from the WAN matching unit, and detects or transmits a Revision-2 Multi-Frequency Code (R2MFC) signal. It converts the voice data transmitted from the function and the switch to IP (Internet Protocol) packet and transmits it to Ethernet or performs the reverse conversion function. The network synchronizer 106 generates 8KHz and 2.048MHz synchronization signals to perform a function of generating a synchronization signal for transmitting serial data between the WAN matching unit, the voice mixing unit, the switch unit, and the processor unit. The processor bus 107 is a parallel bus of address and data and is used to control functions of the WAN matching unit, the voice mixing unit, the switch unit, and the signal processing unit under the control of the processor. This device can use the guide voice and background music as the machine voice. Therefore, it can provide three-way call and background music service through the telephone voice network and the Internet according to the processing of the program. It can be used for analyzing and controlling the service by remote subscribers. In this device, it can be used to control the volume of background music, and PABX (Private Automatic Branch Exchange) using R2MFC signal processing through R2MFC transmission / reception processing of signal processing unit. It is also possible to connect to the system, and it is characterized by the ability to operate as an inter-station signal matching device by a common line signaling method by a processor unit program.

FIG. 2 relates to a digital voice amplification and attenuation method belonging to the function of the voice mixing section 103 in the apparatus of FIG. 1, and detects a DTMF signal transmitted by a caller in the signal processing section 105 to detect the amount of amplification and attenuation. The present invention relates to a method in which amplification and attenuation are performed by varying the digital voice size of a linear code by controlling. The signal 201 input in the figure is converted into a linear code by the linear converter 202 in the manner as shown in FIG. 4, and the code / size separator 203 distinguishes the sign and size bits of the linear code, and amplifies and attenuates it. In the section 204, the size bits are shifted several bits left and right in the manner of FIG. 3 with respect to the size bits, the value bits shifted are added to perform amplification or attenuation as desired, and the code / size combining section ( In 205, the code bit and the amplified / attenuated size bit are combined to generate an amplified or attenuated signal, and the A-Law or Mu-Law coding scheme of G.711 is performed by the nonlinear conversion unit 206 as shown in FIG. By performing the inverse transformation, the amplified or attenuated signal 207 can be obtained. The function block 208 for controlling the volume is composed of a code / size separator 203, an amplification attenuator 204, and a code / size combiner 205.

Amplification results in amplification by + 6dB (Decibel) as the code value is doubled by moving the code of the size part to the left by one bit, and attenuation is caused by moving the code of the size part by one bit to the right. The result is a decrease of 6 dB. The relation is as follows.

Gain (dB) = 20 log (Vout / Vin) = 20 log (2n)

n = (left move), 4, 3, 2, 1, 0, -1, -2, -3, -4,

3dB amplification / attenuation is also possible. 3 dB amplification is the sum of the original signal and the 6 dB signal of the original signal, ie 20 log (1.5) = +3.5 dB, and 3 dB attenuation is the sum of the 6 dB signal of the original signal and the 12 dB signal of the original signal, 20 log (3/4) = Can be implemented at -2.5 dB. In the same way, each of the attenuations obtained by performing attenuation 1, 2, 3, 4, 5 bits, 1/2, 1/4, 1/8, 1/16, 1/32 and amplification 1, 2, 3, Amplification / attenuation control at various intervals is also possible by mixing 2, 4, 8, 16, and 32 amplifications obtained by performing 4 and 5 bits, and the maximum number of shift bits is the number of code bits of the linear code. Up to 12 bits can be shifted when A-Law is converted to linear code, and up to 13 bits when Mu-Law is converted to linear code. If you move many beats to the right, the notes disappear completely. On the other hand, moving many bits to the left can distort the sound as the Maximum Significant Bit (MSB) moves. In this case, it is useful to reduce distortion by allocating signals shifted to the left to the signal whose MSB is 1 and shifted out to the left (Shift Out) to a linear maximum.

3 illustrates amplification and attenuation using a value shifted up to 4 bits to the left and a value shifted up to 5 bits to the right in the method of the amplification attenuator 204 in the function block 208 for controlling the volume of FIG. It shows the code conversion gain when performing. In the figure, the difference between the respective gains can be obtained by adding up to three different values, and about 41 dB can be obtained. By arranging 41 kinds of sizes, a sufficiently fine volume control can be performed. The greater the number of left and right shift bits, the larger the maximum and minimum volume and the finer the level of volume control.

The linear transformation for converting the code of A-Law or Mu-Law into a linear code can be implemented in various ways. 4 shows an example of the conversion using the memory lookup method. The value of the converted linear code has a value from 4096 to 4096 in the case of A-Law. Therefore, 12 bits are used except for the sign bit, and 8159 in the case of Mu-Law. A value exists up to ˜8159, so 13 bits are used except the sign bit. Read the size of the linear code using the lower 7 bits of the 8 bits of A-Law or Mu-Law as the address of the memory. Invert the MSB of the code before conversion and use it as the MSB after conversion. The value assigned to the bit code is used to store 128 quantified values in memory for reading.

Figure 5 shows an example of the conversion of the memory lookup method in the method of converting a linear code into a non-linear code of A-Law or Mu-Law. Converting linear code to A-Law or Mu-Law of G.711 is a bit except for MSB of linear code. High speed processing is possible. After reading the memory, Inverting the MSB of the linear code is used as the MSB of the converted code.

In case of digital processing of voice amplification in the environment using the latest PLD (Programmable Logic Device) technology and VHDL (Very high speed IC Hardware Description Language), it can be applied to various applications and overcomes the disadvantages of analog amplification / attenuation method. In addition, since the digital signal is processed directly, the volume can be controlled by device development and experiment, stable operation and remote control.

Claims (3)

In performing digital speech amplification and attenuation provided with a speech mixing section, A linear converter for converting a digital voice signal state-converted by an A / D converter from a non-linear electrical voice signal to an A-Law to linear and a Mu-Law to linear, A code / size separator for dividing the linearly converted signal with a sign and a size bit of a linear code; A digital voice amplification and attenuation unit for performing various digital voice amplification and attenuation by moving the size bits left or right to optimize the code / size separated signal of the linear code; A code / size combiner which combines the code bit with the various magnitude bits amplified and attenuated to generate as many amplified or attenuated signals as desired; And a non-linear conversion unit for generating an amplified or attenuated nonlinear signal by performing inverse transformation using the A-Law or Mu-Law coding scheme. The method of claim 1, The digital voice amplification and attenuation unit for performing various digital voice amplification and attenuation remotely amplifies or attenuates the volume of the call by amplifying or attenuating the voice in a voice mixing unit step by step according to analysis of a signal such as a dual tone multi-frequency transmitted by a user Digital audio data amplification and attenuation apparatus characterized in that it comprises a remote control to control. In performing digital speech amplification and attenuation provided with a speech mixing section, A linear conversion unit converts a digital voice signal state-converted by an A / D converter from a non-linear electric voice signal to an A-Law to linear and a linear conversion of the electrical voice signal to a Mu-Law to linear; , Dividing the linearly converted signal into a sign and a size bit of a linear code through a sign / size separator; Performing a variety of digital voice amplification and attenuation by moving the magnitude bits left or right through a digital voice amplification and attenuation unit to optimize a code / size separated signal of the linear code; Combining the code bits with the various magnitude bits amplified and attenuated through a code / size combiner to generate as many amplified or attenuated signals as desired; And generating an amplified or attenuated nonlinear signal by performing an inverse transformation through the nonlinear transform unit using the A-Law or Mu-Law coding method.