JPH0324833A - Voice signal transmission system - Google Patents

Voice signal transmission system

Info

Publication number
JPH0324833A
JPH0324833A JP1159255A JP15925589A JPH0324833A JP H0324833 A JPH0324833 A JP H0324833A JP 1159255 A JP1159255 A JP 1159255A JP 15925589 A JP15925589 A JP 15925589A JP H0324833 A JPH0324833 A JP H0324833A
Authority
JP
Japan
Prior art keywords
signal
output
processing
converter
bit
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Pending
Application number
JP1159255A
Other languages
Japanese (ja)
Inventor
Osamu Ichiyoshi
市吉 修
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
NEC Corp
Original Assignee
NEC Corp
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by NEC Corp filed Critical NEC Corp
Priority to JP1159255A priority Critical patent/JPH0324833A/en
Publication of JPH0324833A publication Critical patent/JPH0324833A/en
Pending legal-status Critical Current

Links

Abstract

PURPOSE:To easily attain a privacy talking and to simplify the constitution of a system by transmitting the voice signal in a digital format. CONSTITUTION:In a transmission part, the voice signal is digitized by an A/D converter 2 and separated into frequency information and amplitude information. These pieces of information are arranged in a transmission frame in time division to form a binary data series. This data series is subjected to the enciphering processing and the error correction encoding processing and subjected to prescribed digital modulation by a modulator 11 to be transmitted. In a reception part, the data series is reproduced by digital demodulation, etc., of a demodulator 16, and frame synchronism is settled by detecting a synchronous word to reproduce frequency information and amplitude information, and the compounded audio signal is reproduced from both information. Thus, the privacy talking is easily attained and the constitution of a communication equipment is simplified without enlarging the scale.

Description

【発明の詳細な説明】 (産業上の利用分野) 本発明は音声信号伝送方式に係り、特に移動体通信に好
適な音声信号伝送方式に関する.(従来の技術〉 移動体通信において従来採用されている音声信号伝送方
式としては 例えば第3図に示すものが知られている。
DETAILED DESCRIPTION OF THE INVENTION (Field of Industrial Application) The present invention relates to an audio signal transmission system, and particularly to an audio signal transmission system suitable for mobile communications. (Prior Art) For example, the one shown in FIG. 3 is known as a voice signal transmission method conventionally employed in mobile communications.

第3図において、送信部〈第3図(a))は、ハードリ
ミッタと称される完全圧縮器31と、包路線検波器32
と、FM変調器33と、合波器34と、SSB(単側帯
波)送信機35とを基本的に備える.t’::、受信部
(第3図(b))は、SSB受信1.41と、完全圧縮
器42と、バンドバスフィルタ43と、FM復調器44
と、乗算器45とを基本的に備える.この音声信号伝送
方式は、リンコンベツクス方式と称されているもので、
音声信号を周波数戒分Aと振幅成分Bに分離し2、両者
を別々に送出する(第3図(C)).この時、フエー・
ジングによって変動の恐れある振幅戒分Bを2008Z
程度で狭帯域FM伝送することによって安定化し、品質
改善を図るようにしてある。なお、このリンコンベック
ス方式については、文献「塚田: “VHF帯通信用リ
ンコンペツクス方式の実験と評価“電波時報. 5 ,
 pp2−n <閲56)」に詳述されている. (発明が解決しようkする課H) ところで、移動体通信で用いられる音声信号f云送方式
では、■狭帯域であること、■音声品質が良いこと、■
フェージングに強いこと等の条件の他、■秘話が容易な
こと、■省電力化のため非線形増幅器が使用できること
等の条件も要求される.上述した従来の音声信号伝送方
式は5本質的にアナログ伝送方式であり、■一・−■の
条件は満足するが、■と■の条件を満たしていない.そ
こで,ディジタル化することを考えるのであるが、装置
規模が増大L2移動体通信に適さないものになるεいう
問題がある. 本発明は、このような間Mに鑑みなされたもので、その
目的は,a単な・ディジタル処理で、即ち、装置規模を
増大させずに,秘話が容易に行え、かつ、省電力化が図
れ、以て移動体通信に好適な音声信号伝送方式を提供す
ることにある.(課題を解決するための手段〉 前記目的を達成するために、本発明の音声信号伝送方式
は次の如き構或を有する. 即ち、本発明の音声信号伝送方式は、送信部が、入力し
た音声信号について圧縮処理等を行う前処理部と; こ
の前処理部の出力を所定のサンプル周波数でサンプルし
各サンプル毎にnビットの信号を出力する第1のA/D
変換器と; この第1のA/D変換器の出力を受けて極
性ビットを抽出出力する極性判別器と; 前記第1のA
/D変換器のNサンプル毎の出力について絶対値操作を
しm(m≦n)ビットの信号を出力する振幅検波器と;
 固定ビットパターンからなる同期語を先頭に一定周期
で繰り返す送信フレーム内に前記極性ビットの所定数と
前記mビットの信号とを時分割配置し2進形式のデータ
系列を形成するフレーム合成回路と; このフレーム合
成回路の出力について誤り訂正符号化処理と暗号化処理
を行う送信側ベースバンF信号処理回路と: この送信
側ペースバンド信号処理回路の出力で所定の搬送波をデ
ィジタル変調する変調22と; を!見、受信部が、受
信中間周波数信号に・ノいて準同期復調処理をし受信ベ
ースバンド信号を出力する第1の復調器と; 前記受信
ベースバンド信号をディジタル化する第2のA // 
D変換器と; この第2のA/D変換器の出力について
ディジタル復調処理を行う第2の復調器と; この第2
の復調器の出力について誤り訂正復号化処理と暗号復号
化処理を行う受信側ベースバンド信号妨理回路と: こ
の受信測ベースバンド信号処理回路の出力を受けで前記
同期語の検出によってフレーム同期を確立し前記極性ビ
ットと前記mビットの信号とを分離出力する信号分離回
路と: この信号分離回路が分離出力するmビットの信
号と極性ビットとの乗算を行う乗算器と; この乗算器
の出力をアナログ化するD/AyR換器と: このD/
A変換器の出力について伸張処理,等をし音声信号を再
生出力する後処理部と; を備えていることを特徴とす
るものである. (作 用) 次に、前記の如く楕戒される本発明の音声信号伝送方式
の作用を説明す2る. 送信部では、まず音声信号をディジタル化し、周波数情
報と振幅情報とに分離する.即ち、周波数情報の形成は
極性ビットの抽出のよって行われ、また振幅情報の形或
は第1のA/D変換器のNサンプル毎の出力からmビッ
トの信号を取り出すことによって行われる.即ち、音声
信号の特徴に着目し、極めて単純なディジタル処理によ
って音声信号の符号化を図ったのである。次いで、これ
らの周波数情報と振幅情報を送信フレーム内に時分割配
置し,2進形式のデータ系列を形成する.そして、,二
のデータ系列について暗号化処理と誤り訂正符号化処理
とを行い4所定のディジタル変調を行って送信する。
In FIG. 3, the transmitting section (FIG. 3(a)) includes a perfect compressor 31 called a hard limiter, and an envelope line detector 32.
It basically includes an FM modulator 33, a multiplexer 34, and an SSB (single sideband) transmitter 35. t'::, The reception section (FIG. 3(b)) includes an SSB reception 1.41, a perfect compressor 42, a bandpass filter 43, and an FM demodulator 44.
and a multiplier 45. This audio signal transmission method is called the Rinconvex method.
The audio signal is separated into frequency component A and amplitude component B, and both are sent out separately (Figure 3 (C)). At this time, Hue
2008Z
By transmitting narrowband FM at a certain level, it is stabilized and the quality is improved. Regarding this Rinconvex system, please refer to the document "Tsukada: "Experiment and evaluation of Rinconvex system for VHF band communication" Radio Times. 5,
pp2-n <review 56)''. (Question H that the invention is intended to solve) By the way, in the voice signal transmission system used in mobile communication, there are three requirements: ■ narrow band, ■ good voice quality, and ■
In addition to requirements such as being resistant to fading, other conditions are also required, such as ■ being able to communicate confidentially easily, and ■ being able to use a nonlinear amplifier to save power. The conventional audio signal transmission system described above is essentially an analog transmission system, and although it satisfies the conditions (1) and (2), it does not satisfy the conditions (2) and (3). Therefore, we are considering going digital, but there is a problem of ε, which increases the scale of the equipment and makes it unsuitable for L2 mobile communications. The present invention was developed in view of the above circumstances, and its purpose is to enable simple digital processing, that is, to easily communicate confidential information without increasing the scale of the device, and to save power. The aim is to provide an audio signal transmission system suitable for mobile communications. (Means for Solving the Problems) In order to achieve the above object, the audio signal transmission method of the present invention has the following structure. That is, in the audio signal transmission method of the present invention, the transmitter a pre-processing section that performs compression processing etc. on the audio signal; a first A/D that samples the output of this pre-processing section at a predetermined sampling frequency and outputs an n-bit signal for each sample;
a converter; a polarity discriminator that receives the output of the first A/D converter and extracts and outputs a polarity bit; the first A/D converter;
an amplitude detector that performs absolute value operation on the output of every N samples of the /D converter and outputs an m (m≦n) bit signal;
a frame synthesis circuit that time-divisionally arranges the predetermined number of polarity bits and the m-bit signal in a transmission frame that repeats at a constant period with a synchronization word consisting of a fixed bit pattern at the beginning to form a binary data sequence; a transmitting baseband F signal processing circuit that performs error correction encoding and encryption processing on the output of the frame synthesis circuit; a modulation 22 that digitally modulates a predetermined carrier wave with the output of the transmitting paceband signal processing circuit; ! a first demodulator in which the receiving section performs quasi-synchronous demodulation processing on the received intermediate frequency signal and outputs a received baseband signal; a second demodulator which digitizes the received baseband signal;
a D converter; a second demodulator that performs digital demodulation processing on the output of the second A/D converter;
a receiving side baseband signal jamming circuit that performs error correction decoding processing and encryption decoding processing on the output of the demodulator; a signal separation circuit that establishes and separates and outputs the polarity bit and the m-bit signal; a multiplier that multiplies the m-bit signal separated and outputted by the signal separation circuit by the polarity bit; and the output of the multiplier. A D/AyR converter that converts the D/AyR into analog:
The system is characterized by comprising: a post-processing unit that performs decompression processing, etc. on the output of the A converter and reproduces and outputs the audio signal; and; (Function) Next, the function of the audio signal transmission system of the present invention, which is explained above, will be explained. The transmitter first digitizes the audio signal and separates it into frequency information and amplitude information. That is, the frequency information is formed by extracting the polarity bit, and is also done in the form of amplitude information or by extracting an m-bit signal from the output of every N samples of the first A/D converter. That is, they focused on the characteristics of the audio signal and attempted to encode the audio signal using extremely simple digital processing. Next, these frequency information and amplitude information are time-divisionally arranged within the transmission frame to form a binary data sequence. Then, the second data series is subjected to encryption processing and error correction encoding processing, and is then subjected to predetermined digital modulation and transmitted.

そして、受信部では、ディジタル復調等によって前記デ
ータ系列を再生し.同期語検出によってフレーム同期を
確立して前記周波数情報と振輻悄報を再生し、この両信
号から原音声信号を再生する. 斯くして、本発明の音声信号伝送方式によれば、ディジ
タル形式で音声信号を伝送できるので、秘話を容易に行
うことができ、また送信増幅器に非線形増幅器を用い電
源効率の向上を図ることができる。また、音声信号の符
号化を簡単なディジタル処理によって実現できるので、
通信装置を規模を増大させずにWj素に構成できる.さ
らに、誤り訂正符号化技術を採用しているので、フェー
ジングに強いことは勿論である. 即ち、移動体通信に真に好適な音声信号伝送方式を提供
できるのである. (実 施 例) 以下、本発明の実施例を添付図面を参照して説明する. 第1図は本発明の一実施例に係る音声信号伝送方式を示
す.本実施例方式は、QPSK変調方式を用いたもので
ある. まず、送信部(第1図(a)〉では、音声信号が前処理
部1へ入力する.前処理部1は低域通過ろ波器と圧縮器
を備え、入力した音声信号・にその振幅レベルに応じた
圧縮処理を施してA/D変換器2へ出力する、 Δ,/″′D変換器2唸7タイミング信号発生器6から
供給される所定周波数( IN .jば8KHz)のサ
ンプル信号C,:従って前処理部1の出力をサンプルL
2各サンプル毎に所定nビ・ソト(例えばn−10ビッ
1・〉のサンブIl/値を極性判別器3と振唱検波器4
とへ出力する. 極性判別器3は、A ..,z’ D変換器2の出力サ
ンプル値の極性ビット・9つまり、量上位ビッ!−(M
SB〉を抽出し,それを直披’t.I変換器5へ出力す
る.この極性ビットは周波数情報を示すものである.直
並列変換器5は、タイミング信号発生器6から供給され
る所定周波歎く例えば8 K}!Z)のタイミング信号
に従って極性判別器3の出力ビ・1トを2系列の信号(
P.Q)へ変換する.例えば奇数番目の出力ビットをp
 c言・号系列へ、偶数番目の出力ビットをQ信号系列
へそれぞれ振り分けるのである.極性判別器3の出力デ
ータ速度は8 K bps(bitper secon
d)であるから、この変換された信号系(P,Q)のデ
ータ速度は4Kbpsとなる.これは、フレーム合成回
路7へ出力される.一方、振幅検波器4は、A/D変換
器2のNサンプル毎(倒えば80サンプル毎)の出力サ
ンプル値について絶対値操作をし、即ち、極性ビットが
“1“であれば″0′にする操作をし、このようにした
サンプル値からm(m≦n〉ビット(例えば上位8ビッ
ト)を取り出し、それをビット直列にしてフレーム合成
同路7へ出力する.即ち、振幅情報はサンプル周波数1
00Hzでサンプリングしたものということになり5そ
のデータ速度は8 0 0 bps となる. フレーム合成回路7では、タイミング信号発生器6から
供給されるタイミング信号に従ってP信号系の送信フレ
ーム(第2図(a))とQ信号系の送信フレーム(第2
図(b))とをそれぞれ形戒し、それぞれをベースバン
ド信号処理回F!Il8へ出力する.即ち、1フレーム
は、先頭の同期語(UWP,UWQ)とこれに後続する
データ部とからなり、データ部に振幅情報Aと周波数情
報Fとをこの順序で所定数配置挿入して形成される.具
体的に言えば、振幅検波器4からは8ビットデータが直
列に供給されるので、これを4ビット単位に分割し、一
方をP信号系に他方をQ信号系にそれぞれ割り付ける.
この時、後段のベースバンド信号処理回路8における誤
り訂正符号化処理はこの4ビットについて行われα個の
冗長ビットが付加される.そこで、このフレーム合成回
路7では、振幅情報Aは都合(4+α)ビットの信号と
して配置する.そして、これに後続する周波数情報Fは
、直並列変換器5から予め2系列に分がれて供給される
極性ビット列の対応する信号系における所定数ビットで
構戒される.また、同期語UWPと同UWQは同一ビッ
ト長(例えば40ビット)の固定ビットパターンからな
り、受信側において信号系の識別をするために、そのビ
ットパターンは互いに異ならしめてある. このフレームの長さは、1秒間に例えば15フレーム使
用して振幅情報が400ビット、周波数情報が4000
ビット伝送できるように定めてあるとすれば、同期語は
600ビット伝送される.そして、これが2系列で,伝
送されるから、データ速度は5 0 0 0 sps(
sample per second)ということにな
る.帯域幅はそれ程広くはならないのである. ベースバンド信号処理回路8では、タイミング信号発生
器6からのタイミング信号に従って、フレーム合成回路
7が出力する2系列のデータ列それぞれについて暗号化
処理と誤り訂正符号化処理とを行う。
Then, the receiving section reproduces the data sequence by digital demodulation or the like. Frame synchronization is established by synchronization word detection, the frequency information and vibration information are reproduced, and the original audio signal is reproduced from these two signals. Thus, according to the audio signal transmission system of the present invention, audio signals can be transmitted in digital format, making it possible to easily communicate secret conversations, and using a nonlinear amplifier as a transmission amplifier to improve power efficiency. can. Additionally, audio signals can be encoded using simple digital processing.
Communication equipment can be configured into Wj elements without increasing its scale. Furthermore, since it uses error correction coding technology, it is naturally resistant to fading. In other words, it is possible to provide an audio signal transmission system that is truly suitable for mobile communications. (Example) Examples of the present invention will be described below with reference to the attached drawings. Figure 1 shows an audio signal transmission system according to an embodiment of the present invention. The method of this embodiment uses the QPSK modulation method. First, in the transmitting section (Fig. 1(a)), the audio signal is input to the preprocessing section 1.The preprocessing section 1 is equipped with a low-pass filter and a compressor, and the input audio signal and its amplitude are A sample of a predetermined frequency (IN.j is 8 KHz) supplied from the timing signal generator 6, which is subjected to compression processing according to the level and output to the A/D converter 2. Signal C,: Therefore, the output of preprocessing section 1 is sampled L
2 For each sample, a predetermined sample Il/value of n bits (for example, n-10 bits 1) is determined by the polarity discriminator 3 and the vibration detector 4.
Output to. The polarity discriminator 3 is A. .. , z' The polarity bit 9 of the output sample value of the D converter 2, that is, the upper bit of the quantity! -(M
SB〉 and show it directly. Output to I converter 5. This polarity bit indicates frequency information. The serial-to-parallel converter 5 receives a predetermined frequency, for example 8K}!, supplied from the timing signal generator 6. According to the timing signal of the polarity discriminator 3, the output bit 1 of the polarity discriminator 3 is divided into two series of signals (
P. Convert to Q). For example, if the odd output bit is p
The even-numbered output bits are distributed to the C word/sign sequence and the Q signal sequence, respectively. The output data rate of the polarity discriminator 3 is 8 K bps (bit per second).
d), the data rate of this converted signal system (P, Q) is 4 Kbps. This is output to the frame synthesis circuit 7. On the other hand, the amplitude detector 4 operates the absolute value of the output sample value of the A/D converter 2 every N samples (every 80 samples), that is, if the polarity bit is "1", it is "0' m (m≦n> bits (for example, the upper 8 bits) are extracted from the sample value thus obtained, and are converted into a bit series and output to the frame synthesis circuit 7. In other words, the amplitude information is frequency 1
This means that the data was sampled at 00Hz5, so the data rate is 800 bps. In the frame synthesis circuit 7, according to the timing signal supplied from the timing signal generator 6, a P signal system transmission frame (FIG. 2(a)) and a Q signal system transmission frame (second
Figure (b)) and the baseband signal processing times F! Output to Il8. That is, one frame consists of a synchronization word (UWP, UWQ) at the beginning and a data section following this, and is formed by inserting a predetermined number of amplitude information A and frequency information F in this order into the data section. .. Specifically, since 8-bit data is supplied in series from the amplitude detector 4, this is divided into 4-bit units and one is assigned to the P signal system and the other to the Q signal system.
At this time, error correction encoding processing in the baseband signal processing circuit 8 at the subsequent stage is performed on these four bits, and α redundant bits are added. Therefore, in this frame synthesis circuit 7, the amplitude information A is arranged as a signal of (4+α) bits. The frequency information F that follows this is determined by a predetermined number of bits in the signal system corresponding to the polarity bit string supplied in advance from the serial-to-parallel converter 5 in two series. Further, the synchronization word UWP and the synchronization word UWQ consist of fixed bit patterns of the same bit length (for example, 40 bits), and the bit patterns are made to be different from each other in order to identify the signal system on the receiving side. The length of this frame is, for example, 15 frames per second, 400 bits of amplitude information, and 4000 bits of frequency information.
If it is specified that it can be transmitted in bits, the synchronization word will be transmitted in 600 bits. Since this is transmitted in two series, the data rate is 5000 sps (
sample per second). The bandwidth is not that wide. The baseband signal processing circuit 8 performs encryption processing and error correction encoding processing on each of the two data sequences output from the frame synthesis circuit 7 in accordance with the timing signal from the timing signal generator 6.

このベースバンド信号処理回路8の2系列の出力は対応
する波形整形フィルタ(9P,9Q)、D/A変換器(
IOP,IOQ>を介してQAM変調器11に入力し、
送信搬送波発生器12からの送信搬送波を直交変調した
送信信号として出力される. 次に、受信部(第1図(b))では、受信IF(中間周
波数帯)信号がQAM復調器13へ入力し、ここで受信
搬送波発生器14からの受信搬送波を用いて2系列の信
号が準同期復調される.この2系列の準同期復調信号は
対応するA/D変換器(15P,15Q)にて識別され
、ディジタル復調器16にてベースバンド信号に再生さ
れる。
The two series outputs of this baseband signal processing circuit 8 are output from the corresponding waveform shaping filters (9P, 9Q) and the D/A converter (
input to the QAM modulator 11 via IOP, IOQ>,
The transmission carrier wave from the transmission carrier generator 12 is output as a transmission signal which is orthogonally modulated. Next, in the receiving section (FIG. 1(b)), the received IF (intermediate frequency band) signal is input to the QAM demodulator 13, where the received carrier wave from the received carrier wave generator 14 is used to generate two series of signals. is demodulated quasi-synchronously. These two series of quasi-synchronous demodulated signals are identified by corresponding A/D converters (15P, 15Q) and reproduced into baseband signals by a digital demodulator 16.

なお、タイミング信号再生回路17は、ディジタル復調
器16の再生ベースバンド信号出力に基づきタイミング
信号を再生し、それをA/D変換器(15P,15Q)
、ベースバンド信号処理回路18及び信号分離回路19
へそれぞれ供給している. ベースバンド信号処理回路18は、2系列の再生ベース
バンド信号について誤り訂正復号化処理と暗号復号化処
理をそれぞれ行い、信号分離回路19へ出力する. 信号分離回路19では、フレーム中の同期語(UWP,
UWQ)を検出してフレーム同期を確立し、周波数情報
については2系列のフレーム中からそれぞれ取り出し、
それをそのまま2系列の信号として並直列変換器20へ
出力する.一方、振幅情報については2系列のフレーム
中から取り出して1つの信号列に合成しそれを振幅再生
器21へ出力する. 並直列変換器20は、,送信側の直並列変換器5と逆の
動作を行い、元のビット直列の極性ビット列として乗算
器22の一方の入力に供給する.一方、振幅再生器2l
は、入力したビット直列の振幅情報をビット並列へ変換
し、それを乗算器22の他方の入力に供給する.その結
果、乗算器22からは極性ビットが振幅情報で変調され
た所定ビットパターンの信号が出力される.これはD/
A変換器23にてアナログ化され、後処理部24にて低
域ろ波処理と伸張処理に付され、再生音声信号として出
力される. なお、以上説明した実施例は、QAM方式の適用例であ
るが、PSK (位相変調)方式等、他のディジタル変
調方式であっても同様の考えで実現できることは明らか
である. (発明の効果) 以上説明したように、本発明の音声信号伝送方式によれ
ば、ディジタル形式で音声信号を伝送できるので、秘話
を容易に行うことができ、また送信増幅器に非線形増幅
器を用い電源効率の向上を図ることができる.また.、
音声信号の符号化を簡単なディジタル処理によって実現
できるので、通信装置を規模を増大させずに簡素に構成
できる.さらに、誤り訂正符号化技術を採用しているの
で、フェージングに強いことは勿論である.即ち、移動
体通信に真に好適な音声信号伝送方式を提供できる効果
がある.
Note that the timing signal reproducing circuit 17 reproduces a timing signal based on the reproduced baseband signal output of the digital demodulator 16, and transmits the timing signal to the A/D converter (15P, 15Q).
, baseband signal processing circuit 18 and signal separation circuit 19
They supply each to The baseband signal processing circuit 18 performs error correction decoding processing and encryption decoding processing on the two series of reproduced baseband signals, respectively, and outputs them to the signal separation circuit 19. In the signal separation circuit 19, synchronization words (UWP,
UWQ) is detected to establish frame synchronization, and frequency information is extracted from each of the two series of frames.
The signals are directly output to the parallel-to-serial converter 20 as two series of signals. On the other hand, amplitude information is extracted from two sequences of frames, combined into one signal sequence, and output to the amplitude regenerator 21. The parallel-to-serial converter 20 performs an operation opposite to that of the serial-to-parallel converter 5 on the transmitting side, and supplies it to one input of the multiplier 22 as the original bit series polarity bit string. On the other hand, the amplitude regenerator 2l
converts the input bit-serial amplitude information into bit-parallel information and supplies it to the other input of the multiplier 22. As a result, the multiplier 22 outputs a signal with a predetermined bit pattern in which the polarity bit is modulated with amplitude information. This is D/
The signal is converted into an analog signal by the A converter 23, subjected to low-pass filtering and expansion processing by the post-processing section 24, and output as a reproduced audio signal. Although the embodiment described above is an application example of the QAM method, it is clear that other digital modulation methods such as the PSK (phase keying) method can also be realized using the same concept. (Effects of the Invention) As explained above, according to the audio signal transmission system of the present invention, audio signals can be transmitted in digital format, so confidential conversations can be easily carried out. Efficiency can be improved. Also. ,
Since audio signal encoding can be achieved through simple digital processing, communication equipment can be simply configured without increasing its scale. Furthermore, since it uses error correction coding technology, it is naturally resistant to fading. In other words, it has the effect of providing an audio signal transmission system that is truly suitable for mobile communications.

【図面の簡単な説明】[Brief explanation of drawings]

第1図は本発明の一実施例に係る音声信号伝送方式の構
成ブロック図、第2図は送信フレームの構成図、第3図
は従来例であるリンコンベックス方式の構戒ブロック図
である. 1・・・・・・前処理部、 2・・・・・・A/D変換
器、 3・・・・・・極性判別器、 4・・・・・・振
幅検波器、 5・・・・・・直並列変換器、 6・・・
・・・タイミング信号発生器、7・・・・・・フレーム
合成回路、 8・・・・・・ベースバンド信号処理回路
、 9P,9Q・・・・・・波形整形フィル夕、 IO
P,IOQ・・・・・・D/A変換器、  11・・・
・・・QAM変調器、 12・・・・・・送信搬送波発
生器、l3・・・・・・QAM復調器、 14・・・・
・・受信搬送波発生器、 15P,15Q.・・・・・
・A/D変換器、16・・・・・・ディジタル復調器、
  17・・・・・・タイミング信号再生回路、 18
・・・・・・ベースバンド信号処理回路、 19・・・
・・・信号分離回路、 20・・・・・・並直列変換器
、 21・・・・・・振幅再生器、 22・・・・・・
乗算器、 23・・・・・・D/A変換器、 24・・
・・・・後処理部.
FIG. 1 is a block diagram of the structure of an audio signal transmission system according to an embodiment of the present invention, FIG. 2 is a block diagram of a transmission frame, and FIG. 3 is a block diagram of a conventional Rinconvex system. 1...Preprocessing section, 2...A/D converter, 3...Polarity discriminator, 4...Amplitude detector, 5... ...Series-to-parallel converter, 6...
...Timing signal generator, 7...Frame synthesis circuit, 8...Baseband signal processing circuit, 9P, 9Q...Waveform shaping filter, IO
P, IOQ...D/A converter, 11...
...QAM modulator, 12...... Transmission carrier generator, l3... QAM demodulator, 14...
... Reception carrier wave generator, 15P, 15Q.・・・・・・
・A/D converter, 16...digital demodulator,
17...timing signal regeneration circuit, 18
...Baseband signal processing circuit, 19...
... Signal separation circuit, 20 ... Parallel-serial converter, 21 ... Amplitude regenerator, 22 ...
Multiplier, 23...D/A converter, 24...
...Post-processing section.

Claims (1)

【特許請求の範囲】[Claims] 送信部が、入力した音声信号について圧縮処理等を行う
前処理部と;この前処理部の出力を所定のサンプル周波
数でサンプルし各サンプル毎にnビットの信号を出力す
る第1のA/D変換器とこの第1のA/D変換器の出力
を受けて極性ビットを抽出出力する極性判別器と;前記
第1のA/D変換器のNサンプル毎の出力について絶対
値操作をしm(m≦n)ビットの信号を出力する振幅検
波器と;固定ビットパターンからなる同期語を先頭に一
定周期で繰り返す送信フレーム内に前記極性ビットの所
定数と前記mビットの信号とを時分割配置し2進形式の
データ系列を形成するフレーム合成回路と;このフレー
ム合成回路の出力について誤り訂正符号化処理と暗号化
処理を行う送信側ベースバンド信号処理回路と;この送
信側ベースバンド信号処理回路の出力で所定の搬送波を
ディジタル変調する変調器と;を備え、受信部が、受信
中間周波数信号について準同期復調処理をし受信ベース
バンド信号を出力する第1の復調器と;前記受信ベース
バンド信号をディジタル化する第2のA/D変換器と;
の第2のA/D変換器の出力についてディジタル復調処
理を行う第2の復調器と;この第2の復調器の出力につ
いて誤り訂正復号化処理と暗号復号化処理を行う受信側
ベースバンド信号処理回路と;この受信側ベースバンド
信号処理回路の出力を受けて前記同期語の検出によって
フレーム同期を確立し前記極性ビットと前記mビットの
信号とを分離出力する信号分離回路と;この信号分離回
路が分離出力するmビットの信号と極性ビットとの乗算
を行う乗算器と;この乗算器の出力をアナログ化するD
/A変換器と;このD/A変換器の出力について伸張処
理等をし音声信号を再生出力する後処理部と;を備えて
いることを特徴とする音声信号伝送方式。
The transmitter includes a preprocessor that performs compression processing on the input audio signal; and a first A/D that samples the output of the preprocessor at a predetermined sampling frequency and outputs an n-bit signal for each sample. a converter; a polarity discriminator that receives the output of the first A/D converter and extracts and outputs a polarity bit; and performs absolute value operation on the output of every N samples of the first A/D converter; an amplitude detector that outputs a signal of (m≦n) bits; a predetermined number of polarity bits and the m-bit signal are time-divided within a transmission frame that repeats at a constant period with a synchronization word consisting of a fixed bit pattern at the beginning; a frame synthesis circuit for arranging and forming a binary format data sequence; a transmitting side baseband signal processing circuit for performing error correction encoding processing and encryption processing on the output of this frame synthesizing circuit; and this transmitting side baseband signal processing circuit. a modulator that digitally modulates a predetermined carrier wave with the output of the circuit; a first demodulator in which the receiving section performs quasi-synchronous demodulation processing on the received intermediate frequency signal and outputs a received baseband signal; a second A/D converter for digitizing the band signal;
a second demodulator that performs digital demodulation processing on the output of the second A/D converter; a receiving side baseband signal that performs error correction decoding processing and encryption decoding processing on the output of the second demodulator; a processing circuit; a signal separation circuit that receives the output of the receiving side baseband signal processing circuit, establishes frame synchronization by detecting the synchronization word, and separates and outputs the polarity bit and the m-bit signal; the signal separation circuit; A multiplier that multiplies the m-bit signal that the circuit separates and outputs by the polarity bit; D that converts the output of this multiplier into an analog
1. An audio signal transmission system comprising: a /A converter; and a post-processing unit that performs decompression processing, etc. on the output of the D/A converter and reproduces and outputs the audio signal.
JP1159255A 1989-06-21 1989-06-21 Voice signal transmission system Pending JPH0324833A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP1159255A JPH0324833A (en) 1989-06-21 1989-06-21 Voice signal transmission system

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP1159255A JPH0324833A (en) 1989-06-21 1989-06-21 Voice signal transmission system

Publications (1)

Publication Number Publication Date
JPH0324833A true JPH0324833A (en) 1991-02-01

Family

ID=15689759

Family Applications (1)

Application Number Title Priority Date Filing Date
JP1159255A Pending JPH0324833A (en) 1989-06-21 1989-06-21 Voice signal transmission system

Country Status (1)

Country Link
JP (1) JPH0324833A (en)

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