KR19980050652A - Voice signal storage and playback method of computer using portable communication device - Google Patents

Abstract

The present invention encodes a voice signal in real time using a voice coder (Vocoder) built in a digital portable communication device (hereinafter referred to as a cellular phone) such as a digital cellular phone, a personal mobile communication (PCS), and a GSM. The present invention relates to a method of storing or playing back a voice signal on a computer by real-time encoding.

The present invention relates to a Kim computer using a portable communication device including a microphone for inputting voice information, an analog / digital conversion circuit, a voice encoder implementing a voice encoding algorithm, a CPU, and a transceiver for transmitting and receiving voice information. A method of storing voice data in a storage device, the method comprising: converting an analog voice signal input through a microphone into a digital signal, compressing and encoding a voice signal of one frame of the digitally converted voice signal, and performing compression encoding Packetizing the signal to a computer, checking whether there is an error in the packet received from the mobile phone, and transmitting an NAK signal to the mobile phone if there is an error, and transmitting an ACK signal to the mobile phone if there is an error; Increments the packet counter when it receives a signal, and receives no ACK signal or Checking whether the time of one frame has elapsed; increasing the packet counter if the time of one frame has elapsed; and waiting for a response from the computer if the time of one frame has not elapsed; Checking whether the packet counter transmitted from the controller has increased to 2 or more, and if the check result packet counter has not increased to 2 or more, extracts the received voice frame data from the received packet and stores it in the storage device, and the pegcounter is set to 2 or more. If it is increased, interpolating the frames lost to the previous frame and the currently received frame and storing the interpolated frames in the storage device.

Description

Voice signal storage and playback method of computer using portable communication device

BRIEF DESCRIPTION OF THE DRAWINGS Fig. 1 is a block diagram for explaining sampling, storage and reproduction of an audio signal by a conventional method using a computer.

Fig. 2 is a block diagram for explaining the concept of a speech signal encoding and storage method using a speech method using a computer speech encoder.

3 is a block diagram illustrating the storage of a voice signal in a computer using a mobile phone according to the present invention.

Figure 4 is a block diagram illustrating the reproduction of a voice signal stored in a computer using a mobile phone in accordance with the present invention.

5 is a flowchart illustrating a voice signal compression and storage method performed in a CPU and a voice encoder of a mobile phone.

6 is a flowchart illustrating a method of compressing and storing a voice signal in a computer.

7 is a flowchart for explaining a voice signal decoding and reproduction method performed in a CPU and a voice encoder of a mobile phone.

8 is a flowchart illustrating a method of decoding and reproducing a voice signal in a computer.

* Explanation of symbols for main parts of the drawings

1: microphone 2: analog-to-digital converter

3: Voice encoder 4,9: CPU

5,8: UART (Universal Asynchronous Transceiver) 6,7: Communication port

10: Storage 11: Speaker

The present invention encodes a voice signal in real time using a voice coder (Vocoder) built in a digital mobile communication device (hereinafter referred to as a mobile phone) such as a digital cellular phone, a personal mobile communication (PCS), and a GSM. (real-time encoding) is a method for storing or reproducing voice signals on a computer.

Conventionally, a method of storing a voice signal using a computer has been widely used, and a method of storing a voice signal in a storage device of a computer by using a sound card built in the computer and reproducing it when necessary is common. .

This kind of conventional technology will be described in more detail with reference to FIGS. 1 and 2. 1 is a block diagram illustrating a sampling, storing, and reproducing of a voice signal by embedding a sound card in a computer. As can be seen from FIG. 1, in order to store a voice signal using a computer, a microphone and a speaker for input / output of a sound card and a voice signal are additionally configured in a general computer in hardware. It can be seen that there must be a program for sampling, storing and reproducing the signal. A sound card usually consists of a dedicated chip that processes sound-related data on behalf of the CPU and the circuitry that supports it. The CPU performs data processing related to the whole computer and adjusts the entire operation and operation.The sound card operates separately after receiving a command from the CPU to perform sound-related data processing, and then informs the CPU of the processing result again. After that, it receives another command and starts again.

In general, sound cards can be classified according to the number of bits, sampling rate, MIDI support, and sound synthesis.The amount of data needed to store analog voice signals as digital signals for storage on a computer is It depends on the number and sampling rate. Here, the number of bits refers to the number of bits represented to display sound contents, and 8 bits, 16 bits, and 32 bits are used. An 8 bit sound card means 256 bits (8 powers of 2) of all sound information. Means to modulate with. The sampling rate is also referred to as the recording rate of a computer, but kHz is used as a unit. If the sound card measures 10,000 sound waves per second (called sampling), its sampling rate will be 10 kHz.

Therefore, if the sampling rate is large, the sound quality is excellent, but the data volume is greatly increased. Therefore, in order to store the most voice information with the smallest amount of data, the lowest level of sound quality may be used. The lowest sampling rate is 8,000 Hz and the number of bits per sample is 8 bits. Thus, using the lowest sound quality, the number of beats per second would be 64 kbps, and you would need about 28.8 MB (times of bytes) of storage per hour for recording.

Hereinafter, a process of storing and reproducing a voice signal by the computer will be described. When a program for sampling or recording the voice signal is executed during the storage of the voice signal, the sound card is driven by the CPU, and the sound card is It receives voice information from the microphone, modulates it using bits, and reports the completion to the CPU when the task is completed. The CPU then stores the completed voice information in a storage device such as a hard disk. On the contrary, when a program for reproducing a voice signal is executed in the course of reproducing the voice signal, the CPU receives a voice alarm from a storage device such as a hard disk and instructs the sound card to reproduce the voice signal. The sound card converts the voice information into an analog and outputs it to an external device through an output device such as a speaker. FIG. 2 illustrates a concept of encoding and storing a voice signal using a voice encoder inside a computer. It is a block diagram. In FIG. 2, the speech coder includes a fast arithmetic arithmetic digital signal processing chip (DSP chip) in which a speech coding algorithm is implemented. In FIG. 2, the reason for relying on the additional DSP chip rather than the CPU of the computer is that the speech coding algorithm is large in complexity and computational amount. In other words, if the CPU speed is low, it is impossible to implement the voice encoding algorithm through the computer's CPU. Even if the CPU speed is fast enough to implement it, the voice encoding algorithm is required due to the nature of the CPU. Is too large a load.

Hereinafter, a process of storing and reproducing a voice signal by a computer having a voice encoder will be described. First, the function of the sound card in the process of storing and reproducing a voice signal is the same as that of FIG. However, in FIG. 2, the voice information processed by the sound card is passed to the voice encoder without processing by the CPU so that the voice coder (DSP chip) encodes the data in real time to further reduce the load on the CPU. Here, real-time encoding refers to encoding voice signal data input for a specific time t within t time. The CPU stores the encoded and compressed voice data in a storage device such as a hard disk. On the contrary, in the process of reproducing the voice signal, the voice signal is reproduced by operating in the reverse order of the above process.

However, in a computer device that simply uses a sound card and CPU to store voice signals, although the sound card replaces some functions, the CPU still needs a high-performance CPU and overloads the CPU, making it difficult to use efficiently. There was a problem. In addition, in a computer device having a voice coder using a DSP chip in which a voice encoding algorithm is implemented, a separate sound card, a microphone, and a DSP chip need to be additionally equipped.

The present invention is to solve the above problems, and after connecting the mobile phone to the computer device using a voice encoder and a microphone built in the mobile phone without a separate sound card or microphone without a voice signal compressed to a small amount of data of the computer An object of the present invention is to provide a method of storing and reproducing a storage device.

Voice signal storage method of a computer using a portable communication device of the present invention for achieving the above object, a microphone for inputting voice information, an analog / digital conversion circuit, a voice encoder implemented with a voice encoding algorithm, a CPU; A method of storing voice data in a storage device of a computer using a portable communication device having a transceiver for transmitting and receiving voice information, the method comprising: converting an analog voice signal input through the microphone into a digital signal; Compressing and encoding a voice signal of one frame among the received voice signals, packetizing the compressed and encoded signal to a computer, and checking whether there is an error in the packet received from the mobile phone. Transmitting a NAK signal and transmitting an ACK signal if there is no error; Incrementing a packet counter when receiving an ACK signal from the computer; checking whether one frame has elapsed when the ACK signal is not received from the computer or when a NAK signal is received; Increase the packet counter, wait for a response from the computer if the time of one frame has not yet elapsed, check whether the packet counter transmitted from the mobile phone has increased to two or more, and the check result packet counter If is not increased to 2 or more, extract the voice frame data from the received packet and store it in the storage device, and if the packet counter is increased to 2 or more, interpolating the lost frame with the previous frame and the currently received frame and storing it in the storage device. Characterized in that it comprises a.

In addition, a method of reproducing a voice signal of a computer using the portable communication device of the present invention for achieving the above object includes a speaker, an analog / digital conversion circuit, a voice encoder implemented with a voice encoding algorithm, a CPU, and a transceiver. A method of reproducing voice data from a storage device of a computer using a portable communication device, the method comprising: reading voice data of one frame compressed and stored in the storage device of a computer; And checking whether there is an error in the received packet by receiving the packet transmitted from the computer, and if an error is detected in the received packet as a result of the inspection, checking whether one frame has elapsed, If not detected, transmitting an ACK signal to the computer indicating that a correct packet has been received; As a result of checking whether the time of one frame has elapsed, if the time of one frame has elapsed, it waits for the reception of the next packet. If the time of one frame has not elapsed, it transmits a NAK signal to the computer and waits for the reception of the next packet. And waiting for a response from the mobile phone after transmitting the packet to the mobile phone, and if there is a response, checking whether the signal is an ACK signal, and if it is an ACK signal, reading voice data of a new frame from the storage device; If there is no response from the mobile phone, checking whether the time of one frame has elapsed and waiting for a response from the mobile phone if it has not elapsed, and if it has elapsed, reading voice data of a new frame from the storage device; An ACK signal or a NAK signal indicating whether a packet has been received from the computer Extracting a voice signal from the received packet after transmission, converting the extracted voice signal into an analog signal, and outputting the analog converted voice signal through a speaker. .

The above and other objects and advantages and features of the present invention will become more apparent from the following detailed description of the present invention with reference to the accompanying drawings.

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

Figure 3 is a block diagram for explaining the storage of the voice signal from the mobile phone to the computer according to the present invention, Figure 4 is a block diagram for explaining the reproduction of the voice signal from the computer to the mobile phone according to the present invention.

3 and 4, reference numeral 20 denotes a digital cellular phone, and 30 denotes a computer. In addition, the digital cellular phone 20 converts an analog voice signal input from the microphone 1 for voice input, a speaker 11 for voice output, and an analog voice signal input from the microphone 1 to a digital voice signal, and outputs the speaker 11. An analog-to-digital converter (3) for converting and outputting a digital voice signal into an analog signal; a DSP chip; and a voice encoder (3) for encoding a digitally converted signal by implementing a voice algorithm (Vocoder Algorithm), and a computer connection program. CPU 4, which is operated by a PC, a Universal Asynchronous Transceiver (UART) 5 for transmitting and receiving an encoded signal, and a communication port 6 directly connected to a computer. On the other hand, the computer 30 has a communication port 7 for connecting with the digital mobile phone 20, a universal asynchronous transceiver 8 for transmitting and receiving data with the digital mobile phone, and a mobile station link program for connecting with the digital mobile phone. And a CPU 9 operating by a protocol implemented in a link program and processing the encoded voice information, and a storage device 10 for storing the encoded voice information.

FIG. 5 is a flowchart of processing performed between the CPU in the mobile phone and the voice encoder when storing a voice signal, and FIG. 6 is a flowchart of processing performed in the CPU in the computer when storing the voice signal. Referring to FIGS. 3, 5, and 6, FIG. A method of effectively storing a voice signal in a computer while occupying a small capacity using a voice encoder built in a digital mobile phone will be described.

First, when the mobile phone 20 is turned on, the CPU 4 and the voice encoder 3 in the mobile phone are ready to store a voice signal (S10). When a voice signal is input to the microphone 1 in this state, the analog-to-digital converter 2 converts the input analog signal into a digital signal and outputs it. The speech encoder 3 having a DSP chip on which the speech encoding algorithm is implemented inputs a digitally converted speech signal from the analog / digital converter 2 (S11), and compresses and encodes a speech signal of one frame (S12). In operation S13, the CPU 4 transmits the audio signal of the compressed coded one frame. In this case, a frame usually has a length of about 15-30 ms as a unit of voice signal analysis.

On the other hand, the CPU 4 initializes the packet counter (S14), receives a voice signal of one frame transmitted from the voice encoder 3, and then inserts an error detection code into a packet (S16). A packet is made by adding an error detection code such as a cyclic redundancy checking (CRC) code and information necessary for exchanging data with a computer such as a packet counter to a voice signal frame. The packet is transmitted to the Universal Asynchronous Receiver / Transmitter (LJART) 5 by a connection protocol with a computer implemented by a computer link program. The general purpose asynchronous transceiver 5 transmits the packet to the computer through the communication port 6 according to the RS-232 protocol (S17). The computer receives a packet from the cellular phone input through the communication port 7 and the general purpose asynchronous transceiver 8 (S19), and checks whether there is an error in the received packet (S20). As a result, if there is an error, it transmits a NAK signal to the cellular phone (S21) and returns to step S19 to wait to receive a new packet from the computer, and the error

If there is no SACK signal to the mobile phone (S22). The mobile phone waits for a response from the computer (S18), and upon receiving an ACK signal (S19), increments the packet counter and returns to step S15 to wait for a new frame from the voice encoder 3 (S21). However, if the ACK signal is not received or the NAK signal is received, it is checked whether the time of one frame has elapsed. If the time of one frame has elapsed, the process proceeds to step S25 to increase the packet counter and then the new signal from the voice encoder 3. If the time of one frame has not yet elapsed, the process proceeds to step S18 to wait for a response from the computer.

On the other hand, the computer checks whether the packet counter transmitted from the mobile phone has increased to 2 or more (S26). If the packet counter does not increase to 2 or more, voice frame data is extracted from the received packet and stored in the storage device (S27). In step S19, the process returns to the standby state for receiving a packet from the mobile phone. On the other hand, if the packet counter is increased to 2 or more, the process as described above is repeated by interpolating the frames lost to the previous frame and the currently received frame to the step S27.

The reason for inserting the error correction code and waiting for the ACK / NAK signal from the computer when the packet is touched by the mobile phone in the above-described process is that data may be deformed or lost due to noise during data transmission by the connection between the mobile phone and the computer. This way, when sending data from a mobile phone, the computer can use the error correction code to find out if the data has been deformed, then send an ACK if the data is received correctly, or send a NAK to the mobile phone to request that the previous data be sent again. Will be. The packet counter is used for the following reasons. For example, suppose that one frame of voice signal is 20ms, the CPU of the mobile phone makes it into a packet and transmits it to the computer, and when the ACK signal is received from the computer, it should wait for the next frame. However, if the NAK signal is continuously received from the computer for more than 20ms, the problem occurs. The mobile phone gives up transmitting the current frame and moves on to the next frame. The computer recognizes that the counter has increased by two or more, and stores the unreceived frame in the AED by interpolation using a previous frame and a subsequent frame.

Hereinafter, a method of decoding (extending) and reproducing an audio signal stored in a computer will be described with reference to FIGS. 4, 7 and 8. Fig. 7 is a process flow diagram performed between the CPU in the mobile phone and the voice encoder during decoding and reproducing the voice signal, and Fig. 8 is a process flow chart performed in the CPU in the computer during the decoding and reproduction of the voice signal.

First, when the CPU 9 of the computer is driven by the mobile station link program, the audio data of one frame that is compressed and stored in the storage device 10 is read (S42), and an error detection code is inserted. After the packet is made (S43), the packet is transmitted to the mobile phone (S44).

The mobile phone receives the packet transmitted from the computer (S45) and checks whether an error exists in the received packet (S46). As a result, if an error is detected, it is checked whether the time of one frame has elapsed (S47). If the time of one frame has elapsed, the process proceeds to step S45.

After returning, wait for reception of the next packet. If the time of one frame has not elapsed, a NAK signal is transmitted to the computer, and then the flow returns to step S45 to wait for reception of the next packet. If there is no error in the received packet, an ACK signal is transmitted to the computer indicating that the correct packet has been received (S49).

The computer transmits the packet to the mobile phone and waits for a response from the mobile phone. If there is a response (S50), the computer checks whether the signal is an ACK signal (S51). If the signal is an ACK signal, the computer returns to step S42. Audio data is read, and the flow returns to step S44 if the signal is not an ACK signal. If there is no response from the mobile phone, it checks whether the time of one frame has elapsed. If it has not elapsed, waits for a response from the mobile phone and returns to step S42 to repeat the process.

On the other hand, the mobile phone transmits an ACK signal or NAK signal indicating whether there is an abnormality of the received packet to the computer, extracts the voice signal from the received packet, and transmits the voice signal to the voice encoder 3 (S53). After waiting (S54), the process returns to step S45 to receive a new packet. The voice encoder 3 of the cellular phone receives the audio signal of one frame transmitted from the CPU 4 (S55), decodes it (S53), and transmits it to the digital-to-analog converter 2. The digital / analog converter 2 converts the digital signal into an analog signal and outputs it through the speaker 11 to complete the reproducing operation.

The coding rate of the digital voice signal encoded by the voice encoder 3 varies depending on the type of digital cellular phone. In the case of digital cellular phones currently in service in Korea, a variable coding rate of 8,4,2,1 Kbps is used. 4Kbps or so. In this case, about 1.8MB of storage space is required to store an audio signal for one hour. GSM, which is mainly used in Europe, uses a code rate of 13Kbps, and the required storage space per hour is 5.85MB. Compared with the conventional voice encoder, which requires 28.8MB of storage space per hour, the amount of data can be reduced by 5-7 times, which saves resources and additional devices besides connecting a digital mobile phone to a computer. It can be seen that it is not necessary.

The voice encoder used in the present invention generally includes a decoder and an encoder, and since the decoder has a relatively low complexity and a large amount of calculation compared to the encoder, it may be implemented in software in a computer.

Therefore, when the voice signal is stored, it is possible to follow the method according to the present invention and to reproduce the audio signal using a decoder software and a sound card in a computer without a connection with a mobile phone. In addition, although the communication port is used as the physical layer connecting the mobile phone and the computer in this embodiment, it is also possible to use infrared (IR).

Although the preferred embodiments of the present invention described above only illustrate the most suitable embodiments of the present invention, those skilled in the art will be able to make various modifications, changes, substitutions and additions through the spirit and scope of the present invention. Accordingly, such various modifications, changes, substitutions and additions are intended to be included within the scope of the present invention as set forth in the appended claims.

Claims (4)

A voice is stored in a computer storage device using a portable communication device including a microphone for inputting voice information, an analog / digital conversion circuit, a voice encoder implementing a voice encoding algorithm, a CPU, and a transceiver for transmitting and receiving voice information. A method of storing data, the method comprising: converting an analog voice signal input through the microphone into a digital signal, compressing and encoding a voice signal of one frame of the digitally converted voice signal, and performing the compression coded signal. Packetizing and transmitting the packet to the computer, checking whether there is an error in the packet received from the mobile phone, transmitting an NAK signal to the mobile phone if there is an error, and transmitting an ACK signal if there is no error; Incrementing a packet counter upon receiving a signal, receiving an ACK signal from the computer When the NAK signal is received, checking whether the time of one frame has elapsed; if the time of one frame has elapsed, increasing the packet counter; if the time of one frame has not elapsed, responding from the computer. Waiting; and checking whether the packet counter transmitted from the mobile phone is increased to 2 or more, and extracting the voice frame data from the received packet and storing the packet frame data in the storage device if the packet counter is not increased to 2 or more. And if the paddle counter is increased to 2 or more, interpolating the frames lost to the previous frame and the currently received frame and storing the interpolated frames in the storage device. The method of claim 1, wherein the packet is made by adding an error detection code and information necessary for exchanging data with the computer to the frame. A method of reproducing voice data from a storage device of a computer by using a portable communication device including a speaker, an analog / digital conversion circuit, a voice encoder implementing a voice encoding algorithm, a CPU, and a transceiver, the computer storing Reading voice data of one frame compressed and stored in a device, packetizing the voice data and transmitting the packet to a mobile phone, receiving a packet transmitted from the computer, and checking whether there is an error in the received packet. And checking whether the time of one frame has elapsed when an error is detected in the received packet as a result of the inspection, and if an error is not detected, transmitting an ACK signal indicating that a correct packet is received to the computer; If the time of one frame has elapsed, if the time of one frame has elapsed, Waiting for a scene, if the time of one frame has not elapsed, transmitting a NAK signal to a computer and waiting for reception of the next packet, and after transmitting the packet to a mobile phone, waiting for a response from the mobile phone. Checking whether the signal is an ACK signal, and if it is an ACK signal, reading the audio data of a new frame from the storage device; and if there is no response from the mobile phone, if it has not elapsed by checking whether the time of one frame has elapsed; Waiting for a response from the mobile phone, and if it has passed, reading the voice data of a new frame from the storage device, transmitting an ACK signal or a NAK signal indicating whether there is an error in the packet received from the computer, Extracting a voice signal from the packet, and analyzing the extracted voice signal Converting in the signal, an audio data reproducing method for converting the analog audio signal, characterized in that it comprises a step of outputting through the speaker. 4. The method of claim 3, wherein the packet is made by adding an error detection code and information necessary for exchanging data with the mobile phone to the frame.