KR20050066094A - System of managing a transceiving voice in the mobile communication terminal - Google Patents

Abstract

The present invention relates to a transmission / reception voice processing system in a mobile communication terminal in which a mobile communication terminal records and reproduces transmission and reception voice simultaneously by adding a current modem chip and a simple circuit, and transmits from a microphone amplifier of the mobile communication terminal when a call is performed. A signal synthesizing unit configured to receive the voice and receive the received voice from the receiver amplifier of the mobile communication terminal to perform analog signal synthesis; A codec unit which converts and compresses the signal synthesized by the signal synthesis unit into a digital compression code; It characterized in that it comprises a memory unit for recording the content of the call by storing the signal compressed by the codec unit, the recording time is very long compared to the prior art to prevent the recording of the conversation content is interrupted, recording The mobility of the content can be secured to increase its utility.

Description

System for Managing a Transceiving Voice in the Mobile Communication Terminal

The present invention relates to a transmission and reception voice processing system in a mobile communication terminal, and more particularly, to a transmission and reception voice processing system in a mobile communication terminal in which a mobile communication terminal is capable of simultaneously recording and reproducing transmission and reception voice by addition of a current modem chip and a simple circuit. .

In general, the mobile communication terminal can record only the received voice using a method of storing only the received voice digital signal compressed by QCELP (Qualcomm Code-book Excited Linear Prediction) in memory, that is, the received voice received from the wireless network. The signal is converted into a baseband signal by an RF circuit and extracted into a QCELP format signal by a code division multiple access (CDMA) processor, and the extracted signal is stored in a memory under the control of an ARM core. However, in the case of the method, it is not possible to record the contents of the conversation including the transmission voice, which has a disadvantage in that its utility is low.

Therefore, in order to improve the conventional technique, there is a method of recording not only the received voice but also the transmitted voice using a separate complicated digital signal processing or timing circuit, but this has a very short recording time of 4 to 8 minutes and mobility is not guaranteed. In addition, the current modem chipset has a disadvantage that can not be used as it is.

In other words, it is necessary to add the current mobile communication terminal modem chipset and a simple circuit to record not only the received voice but also the transmitted voice, to guarantee the mobility of the corresponding recording, and to increase its utility.

In order to solve the above disadvantages or necessity, the present invention provides a transmission and reception voice processing system in a mobile communication terminal to simultaneously record and reproduce voice transmission and reception in addition to the current modem chip and a simple circuit in the mobile communication terminal. , Its purpose is.

In addition, the present invention synthesizes the transmitted voice and the received voice in the analog domain by utilizing the mobile communication terminal modem chipset as it is, and compresses and stores, the recording time is very long compared to the prior art, so that the recording of the conversation contents is stopped. To prevent and to ensure the mobility of the recording content to increase its utilization, the purpose is.

In the mobile communication terminal according to an embodiment of the present invention for achieving the above object, the transmission and reception voice processing system receives the transmission voice from the microphone amplifier of the mobile communication terminal when the call is performed and receives the voice from the receiver amplifier of the mobile communication terminal A signal synthesizer which receives the signal and performs analog signal synthesis; A codec unit which converts and compresses the signal synthesized by the signal synthesis unit into a digital compression code; And a memory unit configured to store the signal compressed by the codec unit to record the contents of the call.

Preferably, the signal synthesizing unit attenuates the gain 1/2 so that distortion due to saturation does not occur when synthesizing the transmission voice and the reception voice.

Also, preferably, the codec unit compresses the signal synthesized by the signal synthesizer into an ADPCM format having various formats and sampling rates according to a user's desired voice quality to adjust the storage time within the same capacity of the memory unit. It features.

Further, the codec unit reads and decodes the compressed signal stored in the memory unit when outputting the recording, and outputs it through analog signal processing.

Alternatively, in the mobile communication terminal according to an embodiment of the present invention, the transmission and reception voice processing system includes an A / D converter for converting the signal synthesized in the signal synthesizing unit into a digital signal and applying it to the codec unit; A FIFO for time buffering the signal compressed by the codec unit and the compressed signal read from the memory unit; An interface unit for interfacing the compressed signal transmitted through the FIFO to the memory unit, and for interfacing the compressed signal stored in the memory unit to the FIFO; A D / A converter for converting the signal decoded by the codec into an analog signal; And an audio amplifier for amplifying the analog signal applied from the D / A converter and reproducing the same through a speaker.

The present invention is a memory including an analog signal synthesizing unit by an OP-AMP, a digital compression codec unit, an internal memory or a removable memory card, etc., so that the present invention can be implemented by utilizing a mobile terminal modem chipset as it is. The analog signal synthesizing unit synthesizes the transmitted voice and the received voice using an op amp in the analog domain, and the corresponding codec unit synthesizes the synthesized voice in various formats of digital compression codes (ie, ADPCM (Adaptive Differential Pulse Code). By converting the compressed voice into a built-in memory or a removable memory card by controlling the bus operation of the ARM core in the MSM, and vice versa. Compared to the conventional method, you can record the call as well as the incoming voice. And it gives it to increase the utilization of the long recording time and mobility without the need to modify a separate complex signal processing capabilities of the current modem chipset. Hereinafter, embodiments of the present invention will be described in detail with reference to the accompanying drawings.

In the mobile communication terminal according to an embodiment of the present invention, as shown in FIG. 1, the transmission and reception voice processing system includes an analog signal synthesizing unit 11, an A / D converter 12, an encoder 13, and a FIFO ( 14, an interface unit 15, a memory unit 16, a decoder 17, a D / A converter 18, and an audio amplifier 19.

The analog signal synthesizing unit 11 is a mixer AMP composed of an op amp, and extracts an analog transmission voice from a microphone amplifier of a mobile communication terminal when a mobile communication terminal performs a call between a caller and a called party. After extracting the received voice in analog form from the receiver amplifier of the mobile communication terminal, analog signal synthesis is performed on the extracted transmit voice and the received voice and the synthesized signal is applied to the A / D converter 12. In this case, the gain of the analog signal synthesizing unit 11 does not cause distortion due to saturation, for example, to allow for 1/2 attenuation.

The A / D converter 12 converts an analog signal synthesized from the analog signal synthesizer 11 into a digital signal and applies the converted signal to the encoder 13.

The encoder 13 converts and compresses a digital signal applied from the A / D converter 12 into a digital compression code suitable for a recording quality selected by a user. For example, the encoder 13 compresses the digital signal in various formats of the ADPCM format. The signal is applied to the FIFO 14.

The FIFO 14 performs time buffering on the compressed signal applied from the encoder 13 and transmits the compressed signal to the interface unit 15. The FIFO 14 also transmits the compressed signal applied to the interface unit 15. The buffering is performed and transferred to the decoder 17.

The interface unit 15 interfaces the compressed signal transmitted through the FIFO 14 to the memory unit 16 according to the bus operation control of the ARM core in the MSM of the mobile communication terminal, and also the ARM core in the MSM. The compressed signal stored in the memory unit 16 is read and interfaced to the FIFO 14 according to the control operation of the bus.

The memory unit 16 is a memory (for example, a flash memory, an SRAM, etc.) or a removable memory card built in the mobile communication terminal, and is controlled by bus operation control of an ARM core in the MSM. The compressed signal interfaced from the interface unit 15 is stored to record the call content between the caller and the called party. Also, when the recorded call content is reproduced, the stored compressed signal is controlled by controlling the bus operation of the ARM core in the MSM. Transfer to the interface unit 15.

The decoder 17 decodes the compressed signal transmitted through the FIFO 14 into a general digital signal. For example, the decoder 17 decompresses the compressed signal into a general digital signal and applies it to the D / A converter 18. Do it.

The D / A converter 18 converts the digital signal applied from the decoder 17 into an analog signal and applies it to the audio amplifier 19.

The audio amplifier 19 amplifies an analog signal applied from the D / A converter 18 to reproduce the recorded call contents through a speaker.

Referring to the operation of the transmission and reception voice processing system in a mobile communication terminal according to an embodiment of the present invention.

In the mobile terminal, during the communication between the sender and the called party, the transmitted voice and the received voice are processed from the output of the microphone amplifier of the mobile terminal, and the received voice is extracted from the output of the receiver amplifier of the mobile terminal. Where the output pins are all present in the modem modem chipset.

The extracted voices are all analog signals, synthesized by a mixer amplifier composed of an operational amplifier, and then compressed in an ADPCM format suitable for a recording quality selected by a user and stored in a memory or a memory card in the mobile communication terminal.

On the contrary, the contents of the call recorded in the memory or the memory card are converted into analog by the ADPCM decoder and output through the receiver or the speaker of the mobile communication terminal.

First, the recording operation of the call content in more detail as follows.

First, when a mobile communication terminal performs a call between a caller and a called party, the analog signal synthesizing unit 11, which is a mixer amplifier composed of the op amp, extracts an analog transmission voice from a microphone amplifier of the mobile communication terminal. After extracting the received voice in analog form from the receiver amplifier of the mobile communication terminal, analog signal synthesis is performed on the extracted transmit voice and received voice, and the synthesized signal is applied to the A / D converter 12.

Accordingly, the A / D converter 12 converts the synthesized audio signal of the analog form applied from the analog signal synthesizing unit 11 into a digital signal and applies it to the encoder 13, and the corresponding encoder 13 is The digital signal applied from the A / D converter 12 is converted into a digital compression code suitable for the recording quality selected by the user and compressed.

For example, in the case of an ADPCM encoder, the digital signal applied from the A / D converter 12 is compressed into an ADPCM signal. At this time, various formats of 4 to 8 (bits) of ADPCM are supported. Various sampling rates such as 8, 12.8, 16, and 32 (KHz) are supported, allowing for longer storage times within the same memory capacity depending on the required (i.e. desired) voice quality setting.

Accordingly, the compression-converted signal is stored in the FIFO 14 to perform time buffering through the interface unit 15, and the interface unit 15 receives the compressed signal transmitted through the corresponding FIFO 14. Interface to the memory unit 16 which is a memory (e.g., flash memory, SRAM, etc.) built in the mobile communication terminal or a removable memory card according to the bus operation control of the ARM core in the MSM of the mobile communication terminal. do.

Then, the memory unit 16 stores the compressed signal interfaced from the interface unit 15 by controlling the bus operation of the ARM core in the MSM to record the contents of the call between the caller and the called party. At this time, since the contents of the call are stored in the parallel bus by controlling the bus operation of the ARM core in the MSM, the data transmission speed can be increased.

Secondly, the reproducing operation of the recorded call contents will be described in detail as follows.

First, when reproducing contents of a call between a caller and a called party stored in the memory unit 16, the interface unit 15 receives the compressed signal stored in the memory unit 16 by controlling the bus operation of the ARM core in the MSM. It reads and interfaces with the FIFO 14.

Accordingly, the FIFO 14 performs a time buffering on the compressed signal applied to the interface unit 15, and delivers the compressed signal to the decoder 17, and the corresponding decoder 17 transmits the compressed signal transmitted through the FIFO 14. Decodes into a normal digital signal.

For example, in the case of an ADPCM decoder, the ASPCM signal transmitted through the FIFO 14 is converted into a general digital signal and applied to the D / A converter 18.

Accordingly, the D / A converter 18 converts the digital signal applied from the decoder 17 into an analog signal and applies it to the audio amplifier 19, and the corresponding audio amplifier 19 receives the D / A. The analog signal applied from the converter 18 is amplified to reproduce the recorded call contents through the speaker. In this case, the speaker may be replaced with the receiver of the mobile communication terminal to listen to the contents of the call.

By the operation of the transmission and reception voice processing system in the mobile communication terminal according to the embodiment of the present invention as described above, it is possible to simultaneously record the transmission voice and the reception voice to overcome the limitation of the function that can record only the conventional reception voice It can be implemented by adding a simple mixer amplifier and recording / playback chip without modifying the current MSM, and it can be implemented at low cost without additional design of conventional complex DSP logic. Voice recording quality can be selected by the user to increase the storage time as needed, and recording support on the memory card can be used to increase the storage and usability of stored calls.

As described above, according to the present invention, it is possible to simultaneously record and reproduce the transmitted and received voices in a manner of compressing and storing the received voices in the analog domain by adding the current modem chip and a simple circuit in the mobile communication terminal. Compared to the conventional technology, the recording time is very long to prevent the recording of the conversation contents from being interrupted, and the mobility of the recording contents can be secured to increase its usability.

1 is a block diagram showing a transmission and reception voice processing system in a mobile communication terminal according to an embodiment of the present invention.

Explanation of symbols on the main parts of the drawings

11: analog signal synthesizing unit

12: A / D (Analog to Digital) Converter

13: Encoder 14: FIFO (First In First Out)

15: interface unit 16: memory unit

17: Decoder 18: D / A (Digital to Analog) converter

19: Audio AMP

Claims (5)

A signal synthesizer configured to receive a transmission voice from a microphone amplifier of a mobile communication terminal and to receive analog voice from a receiver amplifier of a mobile communication terminal when performing a call; A codec unit which converts and compresses the signal synthesized by the signal synthesis unit into a digital compression code; Transmitting and receiving voice processing system in a mobile communication terminal, characterized in that it comprises a memory unit for recording the contents of the call by storing the signal compressed by the codec unit. The method of claim 1, And the signal synthesizing unit attenuates the gain 1/2 so that distortion due to saturation does not occur when synthesizing the transmission voice and the reception voice. The method of claim 1, The codec unit compresses the signal synthesized by the signal synthesis unit into an adaptive differential pulse code modulation (ADPCM) format having various formats and sampling rates according to a user's desired voice quality to adjust the storage time within the same capacity of the memory unit. Transmitting and receiving voice processing system in a mobile communication terminal, characterized in that. The method of claim 1, And the codec unit reads and decodes the compressed signal stored in the memory unit when outputting the recording, and outputs the analog signal through the analog signal processing. The method of claim 1, An A / D converter converting the signal synthesized by the signal synthesizing unit into a digital signal and applying the same to the codec unit; A FIFO for time buffering the signal compressed by the codec unit and the compressed signal read from the memory unit; An interface unit for interfacing the compressed signal transmitted through the FIFO to the memory unit, and for interfacing the compressed signal stored in the memory unit to the FIFO; A D / A converter for converting the signal decoded by the codec into an analog signal; Transmitting and receiving voice processing system in a mobile communication terminal, characterized in that it further comprises an audio amplifier for amplifying the analog signal applied from the D / A converter and reproduced through a speaker.