JPS59200543A - Analog voice processor - Google Patents

Analog voice processor

Info

Publication number
JPS59200543A
JPS59200543A JP7410383A JP7410383A JPS59200543A JP S59200543 A JPS59200543 A JP S59200543A JP 7410383 A JP7410383 A JP 7410383A JP 7410383 A JP7410383 A JP 7410383A JP S59200543 A JPS59200543 A JP S59200543A
Authority
JP
Japan
Prior art keywords
circuit
output
signal
frequency
audio
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
JP7410383A
Other languages
Japanese (ja)
Inventor
Hitoshi Odate
大舘 均
Kazuhiro Oguro
一弘 大黒
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.)
Nippon Telegraph and Telephone Corp
Original Assignee
Nippon Telegraph and Telephone 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 Nippon Telegraph and Telephone Corp filed Critical Nippon Telegraph and Telephone Corp
Priority to JP7410383A priority Critical patent/JPS59200543A/en
Publication of JPS59200543A publication Critical patent/JPS59200543A/en
Pending legal-status Critical Current

Links

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04BTRANSMISSION
    • H04B14/00Transmission systems not characterised by the medium used for transmission
    • H04B14/02Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation
    • H04B14/04Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation using pulse code modulation

Landscapes

  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

PURPOSE:To perform radio analog transmission with good voice quality by converting a voice signal to frequencies in a frequency range which is much higher than its original frequency band, and performing one-bit quantization before transmission. CONSTITUTION:A voice signal from a terminal 1 is limited to frequency bands f1-f2 by a BPF2 while a frequency band which exerts no influence upon the voice quality are excluded, and the output of a sine wave oscillator 3 is adjusted to lower than the mean level of the voice signal and inputted to a synthesizer 4 to generate a composite signal. The output of a local oscillator 5 of high frequency f0 is modulated by a balanced modulator 6 with the composite signal and frequency-converted, and a BPF7 extracts (f0+f1)-(f0+f2), which are supplied to a one-bit quantizer 8 for one-bit quantization; and a BPF8 removes higher harmonics generated at the quantizer 8, and a one-bit quantized signal is outputted from an encoder output terminal 10. A decoder multiplies a signal from a terminal 11 by the output of a local oscillator 12 through a mixer 13 and the reproduced voice signal obtained by passing the output of the mixer through an LPF14 and a BPF15 is outputted from a terminal 16.

Description

【発明の詳細な説明】 本発明は、音声入力信号を符号器で符号化して伝送し、
受信側でこれを復号器により復号して取り出すようにし
たアナログ音声処理装置に関するものである。
DETAILED DESCRIPTION OF THE INVENTION The present invention encodes an audio input signal using an encoder and transmits the encoded signal.
The present invention relates to an analog audio processing device in which a decoder decodes and extracts the audio on the receiving side.

従来のこの種の装置では、符号器側で零交差波回路を用
いて音声入力信号を音声周波数帯域のままで1ビット量
子化することが行なわれていたが、これでは復号器側に
おける再生音に雑音が多く、所定の電話品質が得られな
いという欠点があった。
In conventional devices of this type, a zero-crossing wave circuit was used on the encoder side to quantize the audio input signal by one bit in the audio frequency band. The disadvantage was that there was a lot of noise, and the desired telephone quality could not be obtained.

本発明は、上述のような従来技術の欠点を除去するため
になされたものであり、従って本発明の目的は、再生音
に雑音が少なく、所定の電話品質を得ることの可能なア
ナログ音声処理装置を提供することにある。
The present invention has been made in order to eliminate the drawbacks of the prior art as described above, and therefore, an object of the present invention is to provide an analog voice processing method capable of achieving a predetermined telephone quality with less noise in the reproduced sound. The goal is to provide equipment.

本発明の構成の要点は、音声信号をその周波数帯域に比
較して十分高い周波数領域へ周波数変換してから1ビッ
ト量子化を行なうようにした点にあり、それにより良好
な音声品質で無線伝送におけるアナログ伝送を実現する
のに適した音声処理を可能とした。
The key point of the configuration of the present invention is that 1-bit quantization is performed after converting the frequency of the audio signal to a sufficiently high frequency range compared to the frequency band of the audio signal, thereby achieving wireless transmission with good audio quality. This enabled audio processing suitable for realizing analog transmission.

次に図を参照して本発明の一実施例を説明する。Next, an embodiment of the present invention will be described with reference to the drawings.

第1図は本発明の一実施例を示すブロック図である。同
図において、1は音声信号入力端子、2はバンドパスフ
ィルタ(BPF)、3は正弦波発振器、4は合成器、5
はローカル発振器、6は平衡変調器、7はバンドパスフ
ィルタ(BPF)、8はリミ′り回路、9はバンドパス
フィルタ(BPF)、10は符号器出力端子、11は復
号器入力端子、12はローカル発振器、13はミキサ、
14はローパスフィルタ(T、PF)、15はバンドパ
スフィルタ(BPF)、16は再生音出力端子である。
FIG. 1 is a block diagram showing one embodiment of the present invention. In the figure, 1 is an audio signal input terminal, 2 is a band pass filter (BPF), 3 is a sine wave oscillator, 4 is a synthesizer, and 5
is a local oscillator, 6 is a balanced modulator, 7 is a band pass filter (BPF), 8 is a limiting circuit, 9 is a band pass filter (BPF), 10 is an encoder output terminal, 11 is a decoder input terminal, 12 is a local oscillator, 13 is a mixer,
14 is a low pass filter (T, PF), 15 is a band pass filter (BPF), and 16 is a reproduced sound output terminal.

本実施例の動作を以下に詳しく説明する。音声信号入力
端子1に入力した音声信号は、バンドパスフィルタ2で
f 1〜f 2 (例えば、f 1−300 t’lz
The operation of this embodiment will be explained in detail below. The audio signal input to the audio signal input terminal 1 is filtered by the bandpass filter 2 from f 1 to f 2 (for example, from f 1 to 300 t'lz
.

f 2 = 3.4 KHz )の範囲にその周波数帯
域を制限さ弦波発振器3の出力(周波数は例えば2.2
00112)を合成器4にて挿入され合成信号となる。
f 2 = 3.4 KHz)).
00112) is inserted by the combiner 4 to form a composite signal.

該合成信号における正弦波レベルは音声信号の平均レベ
ルよりも低く調整する(例えば20dBのレベル差)。
The sine wave level in the composite signal is adjusted to be lower than the average level of the audio signal (for example, a level difference of 20 dB).

ヂ ローカル発振器5の発振周波数6゜(例えばfo−45
5IGIz )の出力を、平衡変調器6にて前記合成信
号を変調入力として変調し、バンドパスフィルタ7によ
り不要波を除去することにより、周波数帯域(fo−4
−ft )〜(fo+f2)か又は(fo−fl)〜(
fo  f2)へ周波数変換された合成信号を得る(以
下、(fo+f1)〜(fo+f2)の場合について説
明する)。
The oscillation frequency of the dilocal oscillator 5 is 6° (for example, fo-45
The balanced modulator 6 modulates the output of the 5IGIz) using the synthesized signal as a modulation input, and the bandpass filter 7 removes unnecessary waves, thereby converting the frequency band (fo-4
-ft )~(fo+f2) or (fo-fl)~(
A synthesized signal frequency-converted to fo f2) is obtained (hereinafter, the case of (fo+f1) to (fo+f2) will be explained).

周波数変換された合成信号を1ビツト量子器としてのリ
ミッタ回路8により1ビット量子化し、バンドバネフィ
ルタ9にて1ピツト量子化により発生した高調波を除去
し、周波数帯域(fO+fl)〜(fo+f2)の1ビ
ット量子化信号を符号器出力端子10へ出力する。なお
、上述の正弦波信号挿入の目的は音声の無人力時の雑音
抑圧にある。
The frequency-converted composite signal is 1-bit quantized by a limiter circuit 8 serving as a 1-bit quantum device, and harmonics generated by the 1-pit quantization are removed by a band spring filter 9, resulting in a frequency band of (fO+fl) to (fo+f2). A 1-bit quantized signal of 1 is output to the encoder output terminal 10. Note that the purpose of inserting the above-mentioned sine wave signal is to suppress noise when the voice is unattended.

また、1ビツト量子器は例えばリミ′り回路により実現
できるものである。
Further, a 1-bit quantum device can be realized by, for example, a limit circuit.

次に復号器入力端子11に入力した信号を、ローカル発
振器12(発振周波数はローカル発振器5のそれに等し
い)の出力とミキサ13にて乗積し、ローパスフィルタ
14にてミキサ13で発生した高調波を除去し、バンド
パスフィルタ15(通過帯域はバンドパスフィルタ2の
それに等しい)により再生音声信号を得、再生音出力端
子16より出力する。
Next, the signal input to the decoder input terminal 11 is multiplied by the output of the local oscillator 12 (the oscillation frequency is equal to that of the local oscillator 5) in the mixer 13, and the harmonics generated in the mixer 13 are transmitted to the low-pass filter 14. is removed, a reproduced audio signal is obtained by the bandpass filter 15 (pass band is equal to that of the bandpass filter 2), and outputted from the reproduced sound output terminal 16.

第2図は本発明の他の実施例を示すブロック図である。FIG. 2 is a block diagram showing another embodiment of the invention.

同図において、17はバンドパスフィルタ、18は検波
器、19はAGC回路であり他の番号は第1図の実施例
における同一番号のものを示している。
In the figure, 17 is a band pass filter, 18 is a detector, 19 is an AGC circuit, and other numbers are the same numbers as in the embodiment of FIG. 1.

第3図は第2図における合成器4の出力における正弦波
信号に対する音声信号のレベル比とバンドパスフィルタ
ー7の出力における正弦波信号の相対レベルとの関係を
示すグラフである。
FIG. 3 is a graph showing the relationship between the level ratio of the audio signal to the sine wave signal at the output of the synthesizer 4 in FIG. 2 and the relative level of the sine wave signal at the output of the bandpass filter 7 in FIG.

第3図から、正弦波信号のレベルを一定とすれげ、音声
信号のレベルが大きいほど、バンドパスフィルター7の
出力である正弦波信号の再生出力(相対値)が低くなる
ことが理解されるであろう。
From FIG. 3, it can be understood that, assuming the level of the sine wave signal is constant, the higher the level of the audio signal, the lower the reproduction output (relative value) of the sine wave signal that is the output of the bandpass filter 7. Will.

すなわち、バンドパスフィルター7の出力レベルの変化
は音声信号のレベルの変化を表わすものと云うことがで
きる。
In other words, a change in the output level of the bandpass filter 7 can be said to represent a change in the level of the audio signal.

ローハスフィルター4の出力からバンドパスフィルタ1
5により音声信号を得、バンドパスフィルタ17(例え
ば通過帯域2,200±50ITz)により、先に挿入
した正弦波の再生信号を得る。この再生信号のレベルは
音声信号のレベルに逆比例するものであるからこの再生
信号を検波器18で検波し、その検波出力を用いてAG
C回路19の利得を制御し、バンドパスフィルター5よ
り得られご た音声信号振幅成分を更に忠実に再生し、再生音出力端
子161C高品質の再生音を出力する。
From the output of Lohas filter 4 to bandpass filter 1
5 to obtain an audio signal, and a band pass filter 17 (for example, passband 2,200±50 ITz) to obtain a reproduced signal of the previously inserted sine wave. Since the level of this reproduced signal is inversely proportional to the level of the audio signal, this reproduced signal is detected by the detector 18, and the detected output is used to transmit the signal to the AG.
The gain of the C circuit 19 is controlled to more faithfully reproduce the rough audio signal amplitude component obtained from the bandpass filter 5, and high quality reproduced sound is output from the reproduced sound output terminal 161C.

以上説明したように、本発明による音声処理装置を用い
るときは、音声入力信号をその音声周波数帯域より高い
周波数帯に周波数変換してから1ビツト量子化するため
、音声帯域のままで零交差波回路を用いて1ビツト量子
化していた従来装置と比べ、良好な品質で再生音を得る
ことができる。
As explained above, when using the audio processing device according to the present invention, the audio input signal is frequency-converted to a frequency band higher than the audio frequency band and then 1-bit quantized. Compared to conventional devices that use circuits to perform 1-bit quantization, it is possible to obtain reproduced sound with better quality.

また処理された音声波形が定振幅であるため、移動無線
などフェージングによる大きなレベル変動が発生する伝
送路での音声のアナログ伝送用に好適に用いることがで
きる。
Furthermore, since the processed audio waveform has a constant amplitude, it can be suitably used for analog transmission of audio over a transmission line where large level fluctuations due to fading occur, such as in mobile radio.

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

第1図、第2図はそれぞれ本発明の一実施例を示すブロ
ック図、第3図は正弦波信号出力と正弦波信号対音声信
号レベル比の関係を示すグラフ、である。 符号説明 1・・・・・・音声信号入力端子、2・・・・・・バン
ドパスフィルタ、3・・・・・・正弦波発振器、4・・
・・・・合成器。 5・・・・・・ローカル発振器、6・・・・・・平衡変
調器、7・・・・・・バンドパスフィルタ、8・・・・
・・1ビツトit子i。 9・・・・・・バンドパスフィルタ、10・・・・・・
符号器出力端子、11・・・・・・復号器入力端子、1
2・・・・・・ローカル発mW、13・・・・・・ミキ
サ、14・・・・・・ローパスフィルタ、15・・・・
・・バントパスフィルタ、16・・・・・・再生音出力
端子、17・・・・・・バンドパスフィルタ。 18・・・・・・検波器、19・・・・・・AGC回路
代理人 弁理士 並 木 昭 夫 代理人 弁理士 松 崎    清 (ffi4#at)Cl−平ti4’&i4’l’=’
tr?手続補正書 昭和58年6月29日 特許庁長官 若 杉 和 夫 殿 1、事件の表示 特願昭58774103号 2、発明の名称 アナログ音声処理装置3、補正をする
者 事件との関係  特許出願人 住 所 東京都千代田区内幸町1丁目1番6号氏名 (
422) 日本電信電話公社 代表者 真 藤    恒 4、代理人 住 所 東京都港区虎ノ門−丁目5番4号大塚ビル3階 6、補正の対象 図面の5ち第1図および第2図 7、補正の内容 別紙のとおり
FIGS. 1 and 2 are block diagrams showing one embodiment of the present invention, and FIG. 3 is a graph showing the relationship between the sine wave signal output and the sine wave signal to audio signal level ratio. Description of symbols 1...Audio signal input terminal, 2...Band pass filter, 3...Sine wave oscillator, 4...
...Synthesizer. 5...Local oscillator, 6...Balanced modulator, 7...Band pass filter, 8...
...1 bit it child i. 9...Band pass filter, 10...
Encoder output terminal, 11... Decoder input terminal, 1
2...Local source mW, 13...Mixer, 14...Low pass filter, 15...
... Band pass filter, 16 ... Playback sound output terminal, 17 ... Band pass filter. 18...Detector, 19...AGC circuit agent Patent attorney Akio Namiki Agent Patent attorney Kiyoshi Matsuzaki (ffi4#at) Cl-Hirati4'&i4'l'='
tr? Procedural amendment June 29, 1980 Kazuo Wakasugi, Commissioner of the Japan Patent Office1, Indication of the case, Patent Application No. 1983, No. 587741032, Title of the invention: Analog audio processing device 3, Person making the amendment Relationship with the case: Patent applicant Address: 1-1-6 Uchisaiwai-cho, Chiyoda-ku, Tokyo Name (
422) Nippon Telegraph and Telephone Public Corporation Representative Tsune Shinfuji 4, Agent address 3rd floor 6, Otsuka Building, 5-4 Toranomon-chome, Minato-ku, Tokyo, Figure 5 of the drawings subject to amendment, Figures 1 and 2, 7; The details of the amendment are as shown in the attached sheet.

Claims (1)

【特許請求の範囲】 1)音声入力信号を符号器で符号化して伝送した後これ
を復号器で復号して取り出すようにしたアナログ音声処
理装置において、前記符号器は、音声入力信号の音声品
質に影響を与えない周波数帯域において正弦波を挿入す
る第1の回路と、正弦波を挿入された該音声入力信号を
その周波数帯域に比べて十分高い周波数領域へ周波数変
換する第2の回路と、周波数変換された第2の回路から
の出力信号を1ビツト量子化する第3の回路と、量子化
された第3の回路の出力信号からその量子化により発生
した高調波を除去する帯域フィルタとしての第4の回路
を有して成り、該フィルタ出力をもって符号器出力とな
し、前記復号器は、符号器出力を入力されてその周波数
帯域を音声帯域周波数帯に周波数変換する第5の回路と
、該周波数変換出力から再生音を得るための音声帯域フ
ィルタとしての第6の回路を有して成ることを特徴とす
るアナログ音声処理装置。 2)音声入力信号を符号器で符号化して伝送した後これ
を復号器で復号して取り出すようにしたアナログ音声処
理装置において、前記符号器は、音声入力信号の音声品
質に影響を与えない周波数帯域において正弦波を挿入す
る第1の回路と、正弦波を挿入された該音声入力信号を
その周波数帯域に比べて十分高い周波数領域へ周波数変
換する第2の回路と、周波数変換された第2の回路から
の出力信号を1ビツト量子化する第3の回路と、量子化
された第3の回路の出力信号からその量子化により発生
した高調波を除去する帯域フィルタとしての第4の回路
を有して成り、該フィルタ出力をもって符号器出力とな
し、前記復号器は、符号器出力を入力されてその周波数
帯域を音声帯域周波数帯に周波数変換する第5の回路と
、該周波数変換出力から再生音を得るための音声帯域フ
ィルタとしての第6の回路と、前記第5の回路の出力か
ら前記正弦波信号の再生出力を取り出す帯域フィルタと
しての第7の回路と、取り出された正弦波信号の再生出
力を検波する第8の回路と、該検波出力をもって前記第
6の回路の出力である再生音信号の増幅利得を制御して
再生音の品質を高める第9の回路を有して成ることを特
徴とするアナログ音声処理装置。
[Scope of Claims] 1) In an analog audio processing device in which an audio input signal is encoded and transmitted by an encoder and then decoded and extracted by a decoder, the encoder is configured to determine the audio quality of the audio input signal. a first circuit that inserts a sine wave in a frequency band that does not influence a third circuit that quantizes the frequency-converted output signal from the second circuit; and a bandpass filter that removes harmonics generated by the quantization from the quantized output signal of the third circuit. The decoder has a fourth circuit that uses the filter output as an encoder output, and the decoder has a fifth circuit that receives the encoder output and converts the frequency band into a voice band frequency band. , a sixth circuit as an audio band filter for obtaining reproduced sound from the frequency conversion output. 2) In an analog audio processing device in which an audio input signal is encoded and transmitted by an encoder and then decoded and extracted by a decoder, the encoder uses a frequency that does not affect the audio quality of the audio input signal. a first circuit that inserts a sine wave in a frequency band; a second circuit that converts the frequency of the audio input signal into which the sine wave has been inserted into a frequency range that is sufficiently higher than the frequency band; a third circuit for 1-bit quantization of the output signal from the circuit; and a fourth circuit as a bandpass filter for removing harmonics generated by the quantization from the quantized output signal of the third circuit. The decoder includes a fifth circuit that receives the encoder output and converts the frequency band into a voice band frequency band; a sixth circuit as an audio band filter for obtaining reproduced sound; a seventh circuit as a band filter for extracting the reproduced output of the sine wave signal from the output of the fifth circuit; and the extracted sine wave signal. and a ninth circuit that uses the detected output to control the amplification gain of the reproduced sound signal that is the output of the sixth circuit to improve the quality of the reproduced sound. An analog audio processing device characterized by:
JP7410383A 1983-04-28 1983-04-28 Analog voice processor Pending JPS59200543A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP7410383A JPS59200543A (en) 1983-04-28 1983-04-28 Analog voice processor

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP7410383A JPS59200543A (en) 1983-04-28 1983-04-28 Analog voice processor

Publications (1)

Publication Number Publication Date
JPS59200543A true JPS59200543A (en) 1984-11-13

Family

ID=13537514

Family Applications (1)

Application Number Title Priority Date Filing Date
JP7410383A Pending JPS59200543A (en) 1983-04-28 1983-04-28 Analog voice processor

Country Status (1)

Country Link
JP (1) JPS59200543A (en)

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