JPS58215822A - Predictive encoder of voice signal - Google Patents

Predictive encoder of voice signal

Info

Publication number
JPS58215822A
JPS58215822A JP9951482A JP9951482A JPS58215822A JP S58215822 A JPS58215822 A JP S58215822A JP 9951482 A JP9951482 A JP 9951482A JP 9951482 A JP9951482 A JP 9951482A JP S58215822 A JPS58215822 A JP S58215822A
Authority
JP
Japan
Prior art keywords
signal
circuit
predictive
voice signal
filter
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
JP9951482A
Other languages
Japanese (ja)
Inventor
Fumio Sugiyama
文夫 杉山
Makoto Nakamura
誠 中村
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.)
Toshiba Corp
Original Assignee
Toshiba Corp
Tokyo Shibaura Electric Co Ltd
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 Toshiba Corp, Tokyo Shibaura Electric Co Ltd filed Critical Toshiba Corp
Priority to JP9951482A priority Critical patent/JPS58215822A/en
Publication of JPS58215822A publication Critical patent/JPS58215822A/en
Pending legal-status Critical Current

Links

Classifications

    • HELECTRICITY
    • H03ELECTRONIC CIRCUITRY
    • H03MCODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M7/00Conversion of a code where information is represented by a given sequence or number of digits to a code where the same, similar or subset of information is represented by a different sequence or number of digits
    • H03M7/30Compression; Expansion; Suppression of unnecessary data, e.g. redundancy reduction
    • H03M7/3002Conversion to or from differential modulation
    • H03M7/3044Conversion to or from differential modulation with several bits only, i.e. the difference between successive samples being coded by more than one bit, e.g. differential pulse code modulation [DPCM]

Landscapes

  • Engineering & Computer Science (AREA)
  • Theoretical Computer Science (AREA)
  • Filters That Use Time-Delay Elements (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

PURPOSE:To simplify the constitution of a titled device, by constituting a predictive circuit for the encoder encoding a difference signal between an estimated instantaneous value and a present instantaneous value of a lattice type digital filter and an adder summing its outputs, for performing stable predictive processing to a voice signal. CONSTITUTION:A voice signal analyzing circuit obtains a characteristic parameter signal of a voice signal from the past sampling value of the voice signal and inputs the parameter signal to the predictive circuit to obtain the estimated predictive value to the voice signal. This predictive circuit is constituted of the lattice type digital filter 5c and the adder 5d obtaining the total sum of the tap outputs of the filter 5c. A decoded difference signal deltan'' inputted by the filter 5c is transmitted sequentially via adders 21-1-21-8, and the output is transmitted through delay circuits 22-8-22-1 and adders 23-8-23-2 alternately. The output of the adders 21-2-21-8 is applied to multipliers 24-2-24-8 to multiply the autocorrelation coefficient, which is applied to the adders 21-8-21-1 for making the predictive processing stable.

Description

【発明の詳細な説明】 〔発明の技術分野〕 本発明は音声信号を低ビツトレイトで効率良く符号化す
ることのできる音声信号の予測符号化装置に関する。
DETAILED DESCRIPTION OF THE INVENTION [Technical Field of the Invention] The present invention relates to a predictive encoding device for an audio signal that can efficiently encode an audio signal at a low bit rate.

〔発明の技術的背鼠〕[Technical backbone of invention]

音声信号を低ビツトレイトで効率良く予測符号化する装
置として、@1図に示す如く構成された適応型の予測符
号化装置が知られている。
An adaptive predictive coding device configured as shown in Figure 1 is known as a device for efficiently predictive coding an audio signal at a low bit rate.

この装置は標本イヒされた入力音声信号を音声分析回路
1に入力してその特徴パラメータPを抽出すると共に、
遅延回路2を介して減算器3に導びく。上記音声分析回
路1は、例えば該音声信号の過去の複数の標本値から音
声信号の短時間スにクトル包絡線を示すパラメータαや
、ピッチ周波数fp1電力等の特徴ノヤラメータを抽出
するもので、この特徴パラメータ信号は後述する出力回
路4に供給されると共に、予測回路5を構成する予測フ
ィルタ5aに供給され名。
This device inputs a sampled input speech signal to a speech analysis circuit 1 and extracts its characteristic parameter P.
The signal is led to a subtracter 3 via a delay circuit 2. The audio analysis circuit 1 extracts, for example, a parameter α indicating a vector envelope in a short time period of an audio signal, a characteristic parameter such as pitch frequency fp1 power, etc. from a plurality of past sample values of the audio signal. The feature parameter signal is supplied to an output circuit 4, which will be described later, and is also supplied to a prediction filter 5a forming a prediction circuit 5.

一方、前記遅延回路2を介して遅延された讐声信号Sn
は減算器3に導ひかれる。予測回路5が前記特徴・平う
メータ信号を入力して予測推定△ した音声信号に対する推定瞬時値Snと該音声信号Sn
O差信号δ。が求められている。この差信号δ。は符号
化回路6にて符号化され、符号化差信号δ。′として前
記出力回路4に供給されて、前記特徴パラメータ信号P
と共に多重化されて送信されるようになっている。しか
して前記予測回路5は、前記符号化差信号δ。′を復号
回路7を介して復号してなる復号差信号δ。〃と前記推
/\ 定瞬時値(予測値)Sn とを加算器5bにて加算し、
その出力rnを前記予測フィルタ6hK与えている。予
測フィルタ5aは後述するように上記出力γ1と前記特
徴ノJ?ラメータ信号Pとに従△ って推定瞬時値Snを求めている。
On the other hand, the voice signal Sn delayed through the delay circuit 2
is led to subtractor 3. The prediction circuit 5 inputs the feature/flat meter signal and estimates the predicted instantaneous value Sn for the audio signal and the audio signal Sn.
O difference signal δ. is required. This difference signal δ. is encoded by the encoding circuit 6, resulting in an encoded difference signal δ. ' is supplied to the output circuit 4 as the characteristic parameter signal P
It is configured to be multiplexed and transmitted. Therefore, the prediction circuit 5 receives the encoded difference signal δ. ' is decoded via the decoding circuit 7 to produce a decoded difference signal δ. 〃 and the estimate/\ constant instantaneous value (predicted value) Sn are added by an adder 5b,
The output rn is provided to the prediction filter 6hK. The prediction filter 5a uses the output γ1 and the feature J? as described later. The estimated instantaneous value Sn is determined according to the parameter signal P.

さて、このようにして予測符号化されて送信される信号
を受信し、これを復号して信号再生を行う受信再生回路
は、例えば第2図に示す如く構成される。即ち、受信信
号は分配回路8により符号化差信号δ。′と特徴パラメ
ータ信号Pとに分離され、この特徴・ぐラメ′−タ信号
Pは予測回路9の予測フィルタ9aに供給される。また
前記符号化差信号δ。′は復号回路10を介して復号さ
れ、復号差信号δn#として前記予測回路9に供給され
る。この予測回路9により、上記復号差信号δ。〃は加
算器9bに導びかれ、前記△ 予測フィルタ9aが出力する予測信号snに加算され、
再生出力信号Rnとして出力される。この出力信号Rn
が前記音声信号の標本化値snに相当したものとなる。
Now, a receiving and reproducing circuit that receives the predictively encoded and transmitted signal, decodes it, and reproduces the signal is configured as shown in FIG. 2, for example. That is, the received signal is encoded by the distribution circuit 8 into an encoded difference signal δ. ' and a feature parameter signal P, and this feature parameter signal P is supplied to a prediction filter 9a of a prediction circuit 9. Also, the encoded difference signal δ. ' is decoded via the decoding circuit 10 and supplied to the prediction circuit 9 as a decoded difference signal δn#. This prediction circuit 9 generates the decoded difference signal δ. 〃 is guided to the adder 9b and added to the prediction signal sn output from the △ prediction filter 9a,
It is output as a reproduction output signal Rn. This output signal Rn
corresponds to the sampled value sn of the audio signal.

尚、予測フィルタ9aけ上記出力信号Rnを入力し、前
記特徴パラメー、り△ 信号Pに従って予測信号snを作成している。
The above output signal Rn is input to the prediction filter 9a, and a prediction signal sn is created according to the characteristic parameters and the Δ signal P.

このような予測符号化装置と受信再生回路との対によっ
て構成される音声信号のディジタル伝送システムは、例
えば特公昭50−21203号に詳しく紹介されるよう
に、低ビツトレートで音質の良好な音声信号の予測符号
化伝送を行い得る。
A digital transmission system for audio signals consisting of a pair of such a predictive coding device and a receiving/reproducing circuit is capable of transmitting audio signals with low bit rate and good sound quality, as detailed in Japanese Patent Publication No. 50-21203. Predictive coding transmission can be performed.

ところで前記予測回路6を構成する予測フィルタ5&は
、音声信号に対するピッチ周期の予測や過去の複数サン
ダル値から瞬時値の予測を行うものであるが、例えば8
サンプル値から上記瞬時値の線形予測を行う場合、予測
フィルタ5aは一般に第3図に示す如く構成される。即
ち、この予測フィルタ5h、9mは所謂トランス・シー
サル型のものであって、複数段(8段)の遅延回路11
〜+ l I J−2〜ツノ−8を順に介した復号差信
号δ。′の各タップ出方をそれぞれ係数器12 12〜
12−8を介して特徴パラメ−41−2 一夕に基づく係数αl、α鵞〜α8を乗じ、これらの出
力を加算4器13を介して加算合成して出力する如く構
成される。
By the way, the prediction filter 5& constituting the prediction circuit 6 predicts the pitch period of the audio signal and the instantaneous value from a plurality of past sandal values.
When performing linear prediction of the instantaneous value from sample values, the prediction filter 5a is generally constructed as shown in FIG. That is, the prediction filters 5h and 9m are of a so-called transformer-caesal type, and include a plurality of stages (eight stages) of delay circuits 11.
~+ l I Decoded difference signal δ sequentially passed through J-2 to Horn-8. ′ is output from each tap by the coefficient unit 12 12~
12-8, the characteristic parameters 41-2 are multiplied by coefficients αl and α8 to α8 based on the characteristic parameter 41-2, and these outputs are added and combined through an adder 13 and output.

〔背暇技術の問題点〕[Problems with leisure technology]

ところが、上記の如くトランス・ぐ−サル型のフィルタ
を用いて予測フィルタ5aを構成してなる従来の予測回
路5にあっては、上記予測フィルタ5aの伝達関数をF
(2)とした場合、復号差信号δ□〃から信号γ。を作
成する予測回路5゜9自体の伝達関数を1/(]−F(
Z))とすることが必要となる。しかし、このような伝
達関数は、その演算精度の有限性からみて常に安定とな
ることがなく、発振を招来する虞れがある。またα・(
ラメータを計算する計算量は多大である。このような不
具合を解消して安定な予測回路5゜9を構成する為には
、前述したタップ数を増やす等して演獅、精度を高める
ことが必要であり、結局装置の大規模化、消費電力の増
大、装置コストの高騰化等を招き、実用性に問題があっ
た。
However, in the conventional prediction circuit 5 in which the prediction filter 5a is configured using a transformer type filter as described above, the transfer function of the prediction filter 5a is
In the case of (2), the signal γ is obtained from the decoded difference signal δ□〃. The transfer function of the prediction circuit 5゜9 itself that creates 1/(]-F(
Z)). However, such a transfer function is not always stable due to its finite calculation accuracy, and there is a risk of oscillation. Also α・(
The amount of calculation required to calculate the parameters is large. In order to solve these problems and construct a stable prediction circuit 5゜9, it is necessary to increase the number of taps mentioned above to improve performance and accuracy, which ultimately leads to an increase in the scale of the device and This resulted in an increase in power consumption, a rise in device costs, etc., and had problems with practicality.

また受信再生回路にあっては、前記第3図に示す予測フ
ィルタ5aに代えて、所謂)千−コル型の音声合成器で
そのまま構成することは不可能であゲた。
Furthermore, in the reception and regeneration circuit, it is impossible to directly configure a so-called Chi-Col type speech synthesizer in place of the prediction filter 5a shown in FIG. 3.

〔発明の目的〕[Purpose of the invention]

本発明はこのような事情を考慮してなされたもので、そ
の目的とするところは、音声信号に対する予測化処理を
安定に行い得る安価で且つ構成規模の簡単な実用性の高
い音声信号の予測符号化装置Wを提供することにある。
The present invention has been made in consideration of these circumstances, and its purpose is to provide a highly practical audio signal prediction method that is inexpensive, has a simple configuration, and is capable of stably performing prediction processing on audio signals. An object of the present invention is to provide an encoding device W.

〔発明の概要〕[Summary of the invention]

本発明は音声信号の特徴パラメータを用いて上記音声信
号に対する推定瞬時値を予測し、この推定瞬時値と現瞬
時値との差信号を符号化してなる予測符号化装置の予測
回路をラティス型のディジタルフィルタと、このフィル
タのタップ出力の総和を求める加算器とにより構成した
ことを!特徴とするものである。
The present invention uses a lattice-type prediction circuit to predict an estimated instantaneous value of the audio signal using characteristic parameters of the audio signal, and encodes a difference signal between the estimated instantaneous value and the current instantaneous value. It consists of a digital filter and an adder that calculates the sum of the tap outputs of this filter! This is a characteristic feature.

〔発明の効果〕〔Effect of the invention〕

かくして本発明によれば、第3図の予測回路を用いたフ
ィードバック加算処理を行うことなしに、偏自己相関係
数に、を用いて信号のフィルタリング処理を行うので、
単に上記偏自己相関係数をIk、l<1なる条件に設定
するだけで、発振を招くことなしに安定に動作すること
になる。
Thus, according to the present invention, signal filtering processing is performed using the partial autocorrelation coefficient without performing feedback addition processing using the prediction circuit shown in FIG.
Simply setting the partial autocorrelation coefficient to the condition that Ik,l<1 results in stable operation without causing oscillation.

従って常に安定に、しかも低ビツトレートで効率良く音
声信号を予測符号化することが可能と゛ なp、また装
置を小型で簡易に、且つ安価に構成することができる。
Therefore, it is possible to always stably and efficiently predictively encode an audio signal at a low bit rate, and the apparatus can be configured to be small, simple, and inexpensive.

これ故、実用上絶大なる利点・効果が奏せられる。Therefore, tremendous practical advantages and effects can be achieved.

〔発明の実施例〕[Embodiments of the invention]

以下、図面を参照して本発明の一実施例につき説明する
Hereinafter, one embodiment of the present invention will be described with reference to the drawings.

本発明装置は、基本的構成を第1図に示す従来装置と同
じくするものであるが、予測回路5を第4図に示すよう
にラティス型ディジタルフィルタ5Cと加算器5dとに
より構成したことを特徴とするものである。しかしてラ
ティス型ディジタルフィルタ5Cは、入力される復号差
信号δ。〃を加れ器21=、2 L2〜21−8を介し
て順に伝達し、その出力を遅延回路22−8〜22、、
22=と加算器23−8〜23−2を交互に介して順に
伝達する如く構成される。そして、各加算器21−2.
2 L、〜2ノー8の出力は、それぞれ乗算器242+
”−3〜24−8を介して偏自己相関係数kl+に2〜
に7が乗ぜられたのち加算器234.2 L3〜23−
8に供給されるようになっており、また遅延回路22−
8〜22−2.2L1の各遅延出力は乗算器25−8〜
25.、25−。
The device of the present invention has the same basic configuration as the conventional device shown in FIG. 1, except that the prediction circuit 5 is composed of a lattice digital filter 5C and an adder 5d as shown in FIG. This is a characteristic feature. Thus, the lattice type digital filter 5C receives the decoded difference signal δ. 〃 is sequentially transmitted via adders 21=, 2L2 to 21-8, and the outputs are transmitted to delay circuits 22-8 to 22, .
22= and adders 23-8 to 23-2 alternately. And each adder 21-2.
The outputs of 2L, ~2No8 are respectively multiplier 242+
”-3 to 24-8 to the partial autocorrelation coefficient kl+ from 2 to
is multiplied by 7, and then the adder 234.2 L3~23-
8, and the delay circuit 22-
Each delay output of 8 to 22-2.2L1 is output from a multiplier 25-8 to
25. , 25-.

を介して偏自己相関係数に1〜に?+k11が乗ぜられ
たのち前記加算器21−8〜2 L2.2 L、にそれ
ぞれ供給されるようになっている。
to 1 to partial autocorrelation coefficient through? After being multiplied by +k11, the signals are supplied to the adders 21-8 to 2L2.2L, respectively.

しかして、このようなラティス型ディジタルフィルタ5
Cのタップ出力である前記各乗算器25−、、254〜
25−8の各出力が前記加算器5dに供給され、その総
和が求められて推定瞬△ 時値Sn0として出力されるようになっている。
However, such a lattice type digital filter 5
Each of the multipliers 25-, 254-- which is the tap output of C
The respective outputs of 25-8 are supplied to the adder 5d, the sum of which is calculated and outputted as the estimated instantaneous value Sn0.

ラティス型ディジタルフィルタ5Cの各タップ出力X 
H−1・kIの総和を求める加算器5dは、△ なる出力を得るものであり、この出力SnOは△ =Sn として示されることから、前述した予測フィルタ5aと
加算器5bを用いて構成されるフィードツクツクループ
とによって実現される予測回路と全く同様な信号を得る
ことになる。但し、上記関係は、γnを2変換した場合 なる関係が成立することから求められる。
Each tap output of lattice type digital filter 5C
The adder 5d that calculates the sum of H-1·kI obtains an output of △, and since this output SnO is expressed as △=Sn, it is configured using the above-mentioned prediction filter 5a and adder 5b. This results in a signal exactly similar to that of a prediction circuit realized by a feedstock loop. However, the above relationship is obtained from the fact that the relationship holds true when γn is converted by 2.

かくしてこのような予測回路5をラティス型ディジタル
フィルタ5Cと加算器5dとにより構成すれば上記ディ
ジタルフィルタ5Cの偏自己相関係数k をtk、 I
<i に設定するだけで発振のない安定な系を簡易に構
成することが可能となる。従って従来のように演算精度
を高める為に装置構成を大規模化する必要がなく、安価
にして簡易に安定な装置を実現することができ、低ビツ
トレートで効率の高い良好な音声信号の予測符号化が可
能となる。故に、その実用的利点は多大である。
Thus, if such a prediction circuit 5 is configured by a lattice type digital filter 5C and an adder 5d, the partial autocorrelation coefficient k of the digital filter 5C can be expressed as tk, I
By simply setting <i, it is possible to easily construct a stable system without oscillation. Therefore, there is no need to increase the scale of the device configuration in order to improve the calculation accuracy as in the past, and it is possible to easily realize a stable device at low cost, and it is possible to use a low bit rate, high efficiency, and good prediction code for audio signals. It becomes possible to Therefore, its practical advantages are enormous.

尚、本発明は上記実施例に限定されるものではない。例
えば予測回y、sを構成するラティス型ディジタルフィ
ルタ5Cの構成段数は仕様に応じて任意に定めればよい
ものである。またラティス型ディジタルフィルタ5Cの
各偏自己相関係数k はIkiI<1なる条件を満たし
た上で音声信号の特徴パラメータに応じて与えるように
すればよい。要するに本発明はその要旨を逸脱しない範
囲で種々変形して実施することができる。
Note that the present invention is not limited to the above embodiments. For example, the number of stages of the lattice digital filter 5C forming the prediction circuits y and s may be arbitrarily determined according to the specifications. Further, each partial autocorrelation coefficient k of the lattice type digital filter 5C may be given in accordance with the characteristic parameter of the audio signal after satisfying the condition IkiI<1. In short, the present invention can be implemented with various modifications without departing from the gist thereof.

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

第1図は従来の予測符号化装置の概略構成図、第2図は
受信再生回路の構成図、第3図は従来装置における予測
フィルタの構成図、第4図は本発明の一実施例装置に用
いられる予測回路の構成図である。 J・・・音声分析回路、2・・・遅延回路、3・・・減
算回路、5・・・予測回路、5.m・・・予測フィルタ
、5b・・・加算器、5C・・・ラティス型ディジタル
フィルタ、5d・・・加算器、6・・・符号化回路、7
・・・復号回路。 出願人代即人  弁理士 鈴 江 武 彦第1図 第2図 183図
FIG. 1 is a schematic block diagram of a conventional predictive encoding device, FIG. 2 is a block diagram of a receiving and reproducing circuit, FIG. 3 is a block diagram of a predictive filter in the conventional device, and FIG. 4 is a device according to an embodiment of the present invention. FIG. 3 is a configuration diagram of a prediction circuit used for the prediction circuit. J... Voice analysis circuit, 2... Delay circuit, 3... Subtraction circuit, 5... Prediction circuit, 5. m...Prediction filter, 5b...Adder, 5C...Lattice type digital filter, 5d...Adder, 6...Encoding circuit, 7
...Decoding circuit. Takehiko Suzue, Patent Attorney on behalf of the applicant Figure 1 Figure 2 Figure 183

Claims (1)

【特許請求の範囲】[Claims] 音声信号の過去の標本値から上記音声信号の特徴・やラ
メータ信号を求める音声信号分析回路と、上記特徴パラ
メータ信号を入力して前記音声信号に対する推定瞬時値
を推定して求める予測回路と、この推定された音声信号
の推定瞬時値と現瞬時値との差信号を符号化する符号化
回路とを具備し、前記予測回路をラティス型ディジタル
フィルタと、このラティス型ディジタルフィルタのタッ
プ出力の総和を求める加算器とにより構成したことを特
徴とする音声信号の予測符号化装置。
an audio signal analysis circuit that obtains a feature/parameter signal of the audio signal from past sample values of the audio signal; a prediction circuit that inputs the feature parameter signal and estimates and obtains an estimated instantaneous value for the audio signal; an encoding circuit that encodes a difference signal between the estimated instantaneous value and the current instantaneous value of the estimated audio signal; What is claimed is: 1. A predictive coding device for an audio signal, comprising:
JP9951482A 1982-06-10 1982-06-10 Predictive encoder of voice signal Pending JPS58215822A (en)

Priority Applications (1)

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JP9951482A JPS58215822A (en) 1982-06-10 1982-06-10 Predictive encoder of voice signal

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JP9951482A JPS58215822A (en) 1982-06-10 1982-06-10 Predictive encoder of voice signal

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JPS58215822A true JPS58215822A (en) 1983-12-15

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JP9951482A Pending JPS58215822A (en) 1982-06-10 1982-06-10 Predictive encoder of voice signal

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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6584437B2 (en) 2001-06-11 2003-06-24 Nokia Mobile Phones Ltd. Method and apparatus for coding successive pitch periods in speech signal

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS56166632A (en) * 1980-04-21 1981-12-21 Ru Giyuiade Aran Adaptive predicting circuit using differential pcm coding or decoding unit corresponding to lattice filter

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS56166632A (en) * 1980-04-21 1981-12-21 Ru Giyuiade Aran Adaptive predicting circuit using differential pcm coding or decoding unit corresponding to lattice filter

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6584437B2 (en) 2001-06-11 2003-06-24 Nokia Mobile Phones Ltd. Method and apparatus for coding successive pitch periods in speech signal

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