JPS6243223A - Method for adapting adpcm coder to quantizer - Google Patents

Method for adapting adpcm coder to quantizer

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Publication number
JPS6243223A
JPS6243223A JP18177685A JP18177685A JPS6243223A JP S6243223 A JPS6243223 A JP S6243223A JP 18177685 A JP18177685 A JP 18177685A JP 18177685 A JP18177685 A JP 18177685A JP S6243223 A JPS6243223 A JP S6243223A
Authority
JP
Japan
Prior art keywords
signal
quantizer
difference
adaptation
input signal
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
JP18177685A
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Japanese (ja)
Inventor
Shigeo Shinada
品田 重男
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Hitachi Ltd
Original Assignee
Hitachi Ltd
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Filing date
Publication date
Application filed by Hitachi Ltd filed Critical Hitachi Ltd
Priority to JP18177685A priority Critical patent/JPS6243223A/en
Publication of JPS6243223A publication Critical patent/JPS6243223A/en
Pending legal-status Critical Current

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  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

PURPOSE:To prevent a wrong shift to the high-speed adaptation and to attain the adaptive quantization with high quality by detecting the change of an estimating signal as well as the change of a digital input signal supplied to a difference circuit which subtracts the estimating signal from a digitized input signal and suppressing the follow-up of the adapting speed against the increase of the difference signal. CONSTITUTION:For an ADPCM coder, the adapting speed is increased in response to the increase of the estimated difference to produce an unstable answer when the low-speed adaptation is applied to the voice band data signal. Thus, the adaptation is discontinued if the increase of the estimated difference is due to the wrong follow-up. then, the state of a quantizer is held at that time point to prevent the shift to the high-speed adaptation. For instance, an input signal S(j) is supplied to a difference circuit 2 from a terminal 1 at a time point (j) and the difference is obtained between the signal S(j) and an estimated signal Se(j). Thus, a difference signal d(j) is supplied to a quantizer 3. Here, the signal d(j) is scaled by a scale factor Y(j) and controlled to be set at the level of the quantizer 3. Then, the differential PCM quantization value I(j) is delivered to an output terminal 8 and an adverse quantizer 4.

Description

【発明の詳細な説明】 〔発明の利用分野〕 本発明は、音声、音声帯域データ信号、及びトーン信号
等のデータ圧縮の為に用いられる適応差分PCM符号化
器(以下、ADPCM符号化器という)の量子化器適応
化方法に関する。
Detailed Description of the Invention [Field of Application of the Invention] The present invention relates to an adaptive differential PCM encoder (hereinafter referred to as an ADPCM encoder) used for data compression of voice, voice band data signals, tone signals, etc. ) relates to a quantizer adaptation method.

〔発明の背景〕[Background of the invention]

従来のADPCM符号化は、例えば特公表59−500
077号公報やCCITT勧告G721に述べられてい
るように、入力信号が音声の場合には高速の適応化を、
音声帯域データ信号又はトーン信号の場合には低速の適
応化を量子化器に行ない、入力信号に追従する方式をと
っている。
Conventional ADPCM encoding is described in, for example, Japanese Patent Publication No. 59-500.
As stated in Publication No. 077 and CCITT Recommendation G721, when the input signal is audio, high-speed adaptation is required.
In the case of voice band data signals or tone signals, a slow adaptation is applied to the quantizer to follow the input signal.

この従来のADPCM符号化方式では、入力信号と該入
力信号の予測信号との間の誤差が大きくなった場合、量
子化器が低速の適応化状態にあるときに高速の適応化状
態に移行してしまう等の不都合が生じてしまう。
In this conventional ADPCM encoding method, when the error between the input signal and the predicted signal of the input signal becomes large, the quantizer shifts from the slow adaptation state to the fast adaptation state. This may cause inconveniences such as being damaged.

〔発明の目的〕[Purpose of the invention]

本発明の目的は、ADPCM符号化器において、音声帯
域データ信号が入力している時、予測残差の変化に対し
て、誤まった適応化をせず、安定した適応化による量子
化特性の良い量子化器適応化方法を提供する事にある。
An object of the present invention is to prevent erroneous adaptation to changes in prediction residuals and improve quantization characteristics through stable adaptation when a voice band data signal is input to an ADPCM encoder. The purpose is to provide a good quantizer adaptation method.

〔発明の概要〕[Summary of the invention]

ADPCM符号化器においては、音声帯域データ信号に
対して、低速度の適応をしている時、予測残差の増大に
供い適応速度も高速に移行し、不安定な応答となる。
In an ADPCM encoder, when a voice band data signal is adapted at a low speed, the adaptation speed shifts to a high speed as the prediction residual increases, resulting in an unstable response.

そこで本発明では、予測残差の増大が、音声帯域データ
信号の急激な振幅変化によるのか、予測器の追随が不十
分、あるいは誤まった追随によるのかを知シ、後者の場
合には適応化を止め、その時点での量子化器状態を保持
する事によシ高速な適応化に移行しないようにする。こ
れにより安定した適応量子化器になる。
Therefore, in the present invention, it is possible to determine whether the increase in the prediction residual is due to a rapid amplitude change in the voice band data signal, insufficient tracking of the predictor, or incorrect tracking, and in the latter case, adaptive By stopping the quantizer and maintaining the quantizer state at that point, the transition to high-speed adaptation is prevented. This results in a stable adaptive quantizer.

[発明の実施例〕 以下、本発明の一実施例を図面をもとに説明する。第1
図はADPCM符号器のブロック図で、入力端子1、差
回路2、量子化器6、逆量子化器4、加算器5、予測器
6、適応化回路7、出力端子8からなる。
[Embodiment of the Invention] An embodiment of the present invention will be described below with reference to the drawings. 1st
The figure is a block diagram of an ADPCM encoder, which includes an input terminal 1, a difference circuit 2, a quantizer 6, an inverse quantizer 4, an adder 5, a predictor 6, an adaptation circuit 7, and an output terminal 8.

時刻ノにおける入力信号SO1は端子1から差回路2へ
入力し、予測信号5−0)との差がとられ、差信号d(
71が量子化器3へ入力される。そこでは、スケールフ
ァクタYす)でd(71がスケーリングされ、量子化器
3のレベルに入るように調整され、2−ル)・句)が量
子化、符号化され、差動PCM量子化値IO)が出力端
子8及び逆量子化器4へ出力される。逆量子化器4では
、7[1に逆の操作が施され、d(71を量子化した値
dgIjlが出力され、加算器5において予測器6の出
力信号S *(71との和がとられ、入力信号59’l
の再生信号S r(j)が与えられ、予測器6へ入力さ
れる。該信号Sr9°】は次の標本化時刻ノ+1におい
て便われる予測信号Sg(7+1)を生成する為に用い
られる。
The input signal SO1 at time 0 is input to the difference circuit 2 from the terminal 1, and the difference between it and the predicted signal 5-0) is taken, and the difference signal d(
71 is input to the quantizer 3. There, d(71 is scaled and adjusted to fall into the level of quantizer 3, 2-ru) is quantized and encoded by a scale factor Y, and the differential PCM quantized value IO) is output to the output terminal 8 and the inverse quantizer 4. The inverse quantizer 4 performs an inverse operation on 7[1, outputs the value dgIjl obtained by quantizing d(71), and the adder 5 outputs the sum with the output signal S*(71) of the predictor 6. input signal 59'l
A reproduced signal S r (j) is given and input to the predictor 6. The signal Sr9°] is used to generate the prediction signal Sg(7+1) used at the next sampling time +1.

一方、適応化回路7へは、上記1すl 、 5rljl
 。
On the other hand, to the adaptation circuit 7, the above 1sl and 5rljl
.

Souo)が与えられ、次の時刻でのスケールファクタ
Y()+1)の生成に用いられる。第2図において量子
化器ビット数が4ビツトの時の適応化回路7を詳説する
Souo) is given and used to generate the scale factor Y()+1) at the next time. In FIG. 2, the adaptation circuit 7 when the number of quantizer bits is 4 bits will be explained in detail.

第2図における適応化回路は、出力信号1 (71のゆ
らぎを増幅している関数F(・)9.短い時定数の低域
フィルタ10 (例えば411&J a c、低域フィ
ルタを以後LPFと書く。)、長い時定数のLPF 1
3 (例えば1fmzac )、これら2つのL P 
F 10 。
The adaptation circuit in Fig. 2 is a function F(・) that amplifies the fluctuation of the output signal 1 (71). ), long time constant LPF 1
3 (e.g. 1fmzac), these two L P
F10.

l】の出力の差と所定閾値とを比較し、閾値を越える時
は1、さもないときは0を出力する比較器12、該比較
器12の出力を平滑化する短い時定数のL P 113
 (例えば2m5aa )、出力信号11j)からスケ
ールファクタY 91の更新の為の数を発生する関数戦
・114、伝送路績シの影響を減らす為、適応過程に有
限の記憶を導入する為のLP115゜該フィルタ15の
出力を平滑化する為の7.PFlB。
A comparator 12 that compares the difference in the output of [l] with a predetermined threshold and outputs 1 when the threshold is exceeded and 0 otherwise, and a short time constant L P 113 that smoothes the output of the comparator 12.
(e.g. 2m5aa), output signal 11j) to generate a number for updating the scale factor Y91 114, LP115 to introduce finite memory into the adaptation process to reduce the influence of transmission path performance 7. for smoothing the output of the filter 15; PFlB.

これら2つのL P F 15 、16の各出力の算術
平均を前記L P F s3の出力値を重み係数として
とる為の乗算器17 、18、減算器19、加算器20
、出力信号I(7)が過負荷値(II(71+=7) 
 か否かを検出する比較器21、再生信号5rljlの
変化が増加方向か減少方向かを検出する差分回路22、
予測信号5−0)に関する差分回路お、比較器21、差
分回路22゜23.1.PFlBから与えられる信号か
ら、過負荷が入力信号SQlの急速な変化から起こった
か、予測器の能力不足から起こったかを判定する判定器
Uから成る。
Multipliers 17 and 18, subtractor 19, and adder 20 for taking the arithmetic average of the respective outputs of these two L P F 15 and 16 using the output value of L P F s3 as a weighting coefficient.
, the output signal I(7) is the overload value (II(71+=7)
a comparator 21 that detects whether the change in the reproduced signal 5rljl is in an increasing direction or a decreasing direction;
Comparator 21, differential circuit 22, 23.1. It consists of a determiner U which determines from the signal given from PFlB whether the overload has arisen from a rapid change in the input signal SQl or from an insufficiency of the predictor.

まず比較器21において、l l1j)1=7  か否
かが検出され、再生信号5r51、予測信号5sQlは
差分回路22 、23によυ次の差分がとられる;5r
OhSf(>−1)、  5s(jhsacj−1>判
定器スにおいては、過負荷が起こったのは、「α、すl
 < 1/2 、のとき Hn(Srljl−5rCj−1))”qzln(So
ul   5g(j−1))なら予測器の能力不足及は
予測誤シが原因」と判断し、LPFlo、uへその旨出
力する。ここで、すnx は1(J≧O)または−1(
zく0)の値をとる。尚、α、9)はL P 713の
出力信号で、後述するように、入力信号が音声なら1へ
、音声帯域データ信号、及はトーン信号なら0へ近ずく
のでap(7’l < 1/2は後者の入力信号である
事を示す。
First, the comparator 21 detects whether or not l l1j)1=7, and the difference of the υ order is taken between the reproduced signal 5r51 and the predicted signal 5sQl by the difference circuits 22 and 23;
In OhSf(>-1), 5s(jhsacj-1>determiner S, the overload occurs when "α, sl
< 1/2, then Hn(Srljl-5rCj-1))"qzln(So
If ul 5g(j-1)), it is determined that the cause is insufficient ability of the predictor or a prediction error, and outputs that to LPFlo, u. Here, snx is 1 (J≧O) or -1 (
It takes a value of 0). Note that α, 9) is the output signal of L P 713, and as described later, it approaches 1 if the input signal is voice, and approaches 0 if it is a voice band data signal or tone signal, so ap(7'l < 1 /2 indicates the latter input signal.

関数F(す9は次表に従い、I I)’)のゆらぎを増
幅して出力する。
The fluctuation of the function F (S9 is according to the following table, I I)') is amplified and output.

尚、Fσ)=7は過負荷時のゆらぎに大きな重みが与え
られておシ、後段のL P F 10 、11の出力に
大きく影響する事に注意する。L P F 10 、 
Flはそれぞれ短時間平均、長時間平均をとる事に等し
く、それらをdnhzlj) 、 dnhaljlで表
わす。また判定器あの出力が予測器6の追随不備による
過負荷を示している時は、 drLzlj)=ttmz(j−1)、  drpsa
(j)=cLmacj−1)とし、適応を止める。比較
器12ではある閾値Tをもっておシ、dnszlj)と
dmmJ)の差が大きいか、小さいかによ)1又は0を
出力する。(0<Tく1)例えば、 (1−T)dFIKg(7) < dm#(r+ < 
(1+T)ム一〇)→0さもない時        →
1 とすると、音声帯域データ信号のような一定の統計的性
質をもつ定常的な信号では、dmslj)とdma9’
l は互いに近く、比較器はOを出力する。
It should be noted that Fσ)=7 gives a large weight to fluctuations during overload, which greatly affects the outputs of L P F 10 and 11 at the subsequent stage. L P F 10 ,
Fl is equivalent to taking a short-time average and a long-time average, respectively, and these are expressed as dnhzlj) and dnhaljl. Also, when the output of the determiner indicates an overload due to poor tracking of the predictor 6, drLzlj)=ttmz(j-1), drpsa
(j)=cLmacj-1) and stop adaptation. The comparator 12 has a certain threshold value T and outputs 1 or 0 depending on whether the difference between dnszlj) and dmmJ) is large or small. (0<T×1) For example, (1-T)dFIKg(7)<dm#(r+<
(1 + T) Mu 10) → 0 Otherwise →
1, then for a stationary signal with certain statistical properties such as a voice band data signal, dmslj) and dma9'
l are close to each other and the comparator outputs O.

L P 113は該出力を平滑化し、0.1間でゆるや
かな変化をするip (7+ 1 )を与える。
L P 113 smoothes the output and gives ip (7+ 1 ) which varies slowly between 0.1.

関数ス・14はスケールファクタY(7)更新の為の数
を出力するためのもので、L P F 15を通してY
v(7)を出力する。仁のロバスト化された高速スケー
ルファクタYu(j)はL P F 16で平滑化され
、音声帯域データのような信号に都合のよいスケールフ
ァクタYa(jlを与える。これらYμIj)、Ya(
jlの算術平均を前記αp (、)’ + 1 )の重
みで求めY()+1)を得る; 1’(7+1 ) −4pO+1 )Yu(71+ (
1−αpO”1 )Ya(71本実施例によれば、Cく
1TT、V29勧告の9.6kApz  のような振幅
変化の大きい音声帯域データが入力した場合に発生する
予測器6の追随不備による過負荷時においても、drp
sa5)、dnha(j)は更新されず、従って、α、
0)は1の方に動かないため、Y (7’lは低速のス
ケール7アクタY−0)のままとどま)、不安定な適応
化とならない効果がある。
The function S.14 is for outputting the number for updating the scale factor Y (7), and it is used to output the number for updating the scale factor Y (7).
Output v(7). Jin's robust fast scale factor Yu(j) is smoothed with L P F 16 to give a scale factor Ya(jl, which is convenient for signals such as voice band data. These YμIj), Ya(
Calculate the arithmetic mean of jl with the weight of αp (,)' + 1) to obtain Y()+1); 1'(7+1) -4pO+1)Yu(71+ (
1-αpO"1)Ya (71According to this embodiment, this is due to the tracking defect of the predictor 6 that occurs when audio band data with large amplitude changes such as 9.6kApz of C1TT and V29 recommendations is input. Even during overload, drp
sa5), dnha(j) is not updated, therefore α,
0) does not move toward 1, Y (7'l remains the slow scale 7 actor Y-0), which has the effect of not resulting in unstable adaptation.

〔発明の効果〕〔Effect of the invention〕

本発明によれば、音声帯域データ信号、あるいはトーン
信号のような、はぼ一定の統計的性質を有する定常的な
信号が入力している時、予測器の追随不備による残差信
号の増大を、入力信号の急速な振幅変化と考え、誤まっ
て高速な適応化へ移行する事が防げられるので、上記の
ような信号に対して高品質の適応量子化が期待できる。
According to the present invention, when a stationary signal with almost constant statistical properties is input, such as a voice band data signal or a tone signal, the increase in the residual signal due to the tracking defect of the predictor is prevented. , it is possible to prevent erroneous transition to high-speed adaptation by considering rapid amplitude changes of the input signal, so high-quality adaptive quantization can be expected for the above-mentioned signals.

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

第1図は本発明の一実施例に係るADPCM符号化器の
ブロック図、第2図は、第1図中の適応化回路の詳細ブ
ロック図である。
FIG. 1 is a block diagram of an ADPCM encoder according to an embodiment of the present invention, and FIG. 2 is a detailed block diagram of the adaptation circuit in FIG. 1.

Claims (1)

【特許請求の範囲】[Claims] 標本化時刻毎に入力するデジタル化入力信号から該入力
信号の予測信号を引き去る差回路の出力信号を適応量子
化器で量子化して差動PCM信号に変換するADPCM
符号化器の、上記適応量子化器の適応速度を入力信号に
応じて制御する量子化器適応化法において、上記差回路
に入力するデジタル入力信号の変化と予測信号の変化と
を検知し、適応速度が低速の場合には、上記二つの変化
を見て、これらの変化方向が逆なために起こる差信号増
大に対しては適応速度の追随を抑え、適応量子化器の状
態を保持する事を特徴とするADPCM符号化器の量子
化器適応化方法。
ADPCM in which the output signal of a difference circuit that subtracts the predicted signal of the input signal from the digitized input signal input at each sampling time is quantized by an adaptive quantizer and converted into a differential PCM signal.
In the quantizer adaptation method of controlling the adaptation speed of the adaptive quantizer of the encoder according to the input signal, detecting a change in the digital input signal input to the difference circuit and a change in the prediction signal, When the adaptation speed is slow, the above two changes are observed, and when the difference signal increases due to the opposite direction of change, the adaptation speed is suppressed from following, and the state of the adaptive quantizer is maintained. A quantizer adaptation method for an ADPCM encoder, characterized in that:
JP18177685A 1985-08-21 1985-08-21 Method for adapting adpcm coder to quantizer Pending JPS6243223A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP18177685A JPS6243223A (en) 1985-08-21 1985-08-21 Method for adapting adpcm coder to quantizer

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP18177685A JPS6243223A (en) 1985-08-21 1985-08-21 Method for adapting adpcm coder to quantizer

Publications (1)

Publication Number Publication Date
JPS6243223A true JPS6243223A (en) 1987-02-25

Family

ID=16106680

Family Applications (1)

Application Number Title Priority Date Filing Date
JP18177685A Pending JPS6243223A (en) 1985-08-21 1985-08-21 Method for adapting adpcm coder to quantizer

Country Status (1)

Country Link
JP (1) JPS6243223A (en)

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