KR930024408A

KR930024408A - LSP Vocoder

Info

Publication number: KR930024408A
Application number: KR1019920009166A
Authority: KR
Inventors: 조용덕; 김홍국; 공병구
Original assignee: 정용문; 삼성전자 주식회사
Priority date: 1992-05-28
Filing date: 1992-05-28
Publication date: 1993-12-22
Also published as: KR0175248B1

Abstract

음성전송 시스템에 있어서, 보코더에 관한 것으로 특히 LSP 음성 부호화 기법을 사용하여 전송캐넌을 확장하여 전송하는 LSP보코더에 관한 것이다.In a voice transmission system, the present invention relates to a vocoder, and more particularly, to an LSP vocoder for extending and transmitting a transmission canon using an LSP speech encoding technique.

64kbps의 μ-law PCM신호를 8kbps이하의 데이타로 압축하는 부호화기(100)와, 상기 부호화기(100)에서 압축된 8kbps이하의 데이타를 64kbps의 μ-law 신호로 신장하는 복조화기 (200)로 구성하여 채널당 전송률은 감소시켜 트렁크 증설없이 트렁크당 채널수를 증가시키고 채널 사용료를 줄인다.The encoder 100 compresses a 64 kbps μ-law PCM signal into data of 8 kbps or less, and the demodulator 200 extends 8 kbps or less data compressed by the encoder 100 into a 64 kbps μ-law signal. As a result, the transmission rate per channel is reduced, thereby increasing the number of channels per trunk and reducing the channel usage fee without additional trunks.

Description

LSP Vocoder

본 내용은 요부공개 건이므로 전문내용을 수록하지 않았음Since this is an open matter, no full text was included.

제2도는 본 발명에 따른 음성전송 시스템의 블럭 구성도, 제3도는 본 발명에 따른 LSP보코더의 블럭 구성도, 제4도는 제2도중 음성 분석부(101)에서 음성을 분석하고 소프트웨어 모듈 구성도.2 is a block diagram of a voice transmission system according to the present invention, FIG. 3 is a block diagram of an LSP vocoder according to the present invention, and FIG. 4 is a block diagram of the voice analysis unit 101 in FIG. .

Claims

In the LPS vocoder of the voice transmission system, the encoding unit 100 compresses a 64kbps μ-law PCM signal into 8kbps or less data, and the 8kbps or less data compressed by the encoder 100 is 64kbps μ-law signal. LSP vocoder of the voice transmission system, characterized in that consisting of a decoding unit 200 extending to.

2. The encoder 100 of claim 1, wherein the encoder 100 analyzes a speech by calculating an LSP coefficient pitch, punishment / unvoiced punctuation, and an average amplitude of a frame from a 64 kbps μ-law PMC signal. A tone detector 102 for detecting the type of DTMF or call progress tone signal from a 64 kbp μ-law PCM signal, amplitude, pitch, LSP coefficient, voice / unvoiced flag calculated by the voice analyzer 101 and the tone detector A quantization unit 103 for quantizing the DTMF or the call progress tone signal determined in step 102 by 80 bits and a quantized signal in the quantization unit 103 are classified into three types of codes, respectively, to form a CRC code and a packet. LSP vocoder of the voice transmission system, characterized in that consisting of a packetization unit 104 for the speech.

According to claim 1, wherein the decoder 200 inputs the packetized signal from the packetizer 104 to separate the amplitude, pitch, DTMF information, LSP coefficients and generates CRC codes according to three data types Then, by comparing the secondary value, the unpacketizer 201 and the unpacketized signal are inputted by the unpacketizer 201 to invert the amplitude, pitch, LSP coefficient, DTMF, or call progress tone signal. The inverse quantization unit 202 and the inverse quantization unit 202 input quantized inverse quantized voice signals, and sound sources are modeled as pulses and random noises according to pitch periods, respectively, according to the voiced / unvoiced sound. A speech synthesizer 203 for generating speech synthesis by the sound source and a dequantized tone signal from the inverse quantizer 202, and searching for a corresponding frequency analyzed as DTMF or a call progress tone. Corresponding The LSP vocoder of the voice transmission system, characterized in that it comprises a tone generating unit 204 for generating a tone signal by calling a sign function.

※ Note: The disclosure is based on the initial application.