JPH0870285A - Voice decoder - Google Patents

Voice decoder

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Publication number
JPH0870285A
JPH0870285A JP7173884A JP17388495A JPH0870285A JP H0870285 A JPH0870285 A JP H0870285A JP 7173884 A JP7173884 A JP 7173884A JP 17388495 A JP17388495 A JP 17388495A JP H0870285 A JPH0870285 A JP H0870285A
Authority
JP
Japan
Prior art keywords
section
prediction coefficient
voice
signal
detection flag
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.)
Granted
Application number
JP7173884A
Other languages
Japanese (ja)
Other versions
JP3593183B2 (en
Inventor
Ichiro Matsumoto
一郎 松本
Osamu Watanabe
治 渡辺
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.)
Kokusai Electric Corp
Original Assignee
Kokusai Electric Corp
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Filing date
Publication date
Application filed by Kokusai Electric Corp filed Critical Kokusai Electric Corp
Priority to JP17388495A priority Critical patent/JP3593183B2/en
Publication of JPH0870285A publication Critical patent/JPH0870285A/en
Application granted granted Critical
Publication of JP3593183B2 publication Critical patent/JP3593183B2/en
Anticipated expiration legal-status Critical
Expired - Fee Related legal-status Critical Current

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  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

PURPOSE: To improve a problem of a pseudo background noise for a silence interval of receiver side causing unnatural sound quality when a sound detection flag is outputted by mistake for the silence interval from the sender side in the receiver side voice decoder for the adaptive differential PCM coding system communication. CONSTITUTION: An auxiliary controller 6 is provided between a controller 5 receiving a sound detection flag (b) separated from a reception demodulation signal (a) to provide an output of a signal (e) representing discrimination of sound/silence and a reset signal (d) at a change point of the sound/silence and a predict,ion coefficient storage device 7 extracting a prediction coefficient (g) from an ADPCM decoder 4 decoding a coded signal (c), calculating an average for each frame, storing and updating it and providing an output of a stored value (h) just before the change from the silence interval into the sound interval. The auxiliary controller 6 outputs continuously a flag (f) representing the silence interval for nearly 5 frames even when the input flag (e) indicates the change of the silence interval into the sound interval so that the sound quality of pseudo background noise generated and decoded by the decoder 4 for the silence interval is not disturbed.

Description

【発明の詳細な説明】Detailed Description of the Invention

【0001】[0001]

【産業上の利用分野】本発明は音声復号装置に関するも
ので、特に符号化方式に適応差分PCM(ADPCM)
方式を適応した受信信号を復号する音声復号装置に関す
るものである。
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a speech decoding device, and more particularly to an adaptive differential PCM (ADPCM) coding system.
The present invention relates to a voice decoding device that decodes a received signal that is compatible with the method.

【0002】[0002]

【従来の技術】音声による通信が行われている場合、ど
ちらか一方が発声している時間率は約35%であるとい
われている。近年、個人主体の通信であるパーソナルコ
ミュニケーションがその範囲を拡大してきている。そこ
では携帯に便利な端末を利用した音声通信が主体であ
る。このような携帯端末に要求される事項として、第一
に、コードレス化が挙げられる。第二に、携帯に便利な
ように電池が使用され長時間にわたる使用に耐える必要
があるため回路消費電力の低減化が要求されている。
2. Description of the Related Art When voice communication is performed, it is said that the time rate during which either one is uttering is about 35%. In recent years, the range of personal communication, which is a communication mainly based on individuals, has been expanding. There, mainly voice communication using a portable terminal is used. As a matter required for such a mobile terminal, firstly, there is a cordless method. Secondly, it is required to reduce the circuit power consumption because a battery is used for convenience of carrying and it is required to endure long-term use.

【0003】回路消費電力の従来以上の低減方法とし
て、音声の発声時間率に着目し、発声している時のみ送
信回路を動作させ、その他の送信時間は回路を休止状態
にする方法がある。このような技術を実現するためには
送信側に音声検出機能を設けて不連続送信装置を付加す
れば良い。その場合に問題となるのは受信側である。す
なわち、受信側では再生音声が断続するために非常に不
愉快な音声になる。この原因は音声を伝送しているとき
は音声に背景雑音が重畳されているが、音声が伝送され
ないときは背景雑音も伝送されず、背景雑音が音声信号
の持続の時のみ変調されて伝送されるためであることが
知られている。
As a method of reducing the circuit power consumption more than ever before, there is a method of paying attention to the utterance time rate of voice, operating the transmission circuit only when uttering, and putting the circuit in a rest state during the other transmission time. In order to realize such a technique, a voice detection function may be provided on the transmission side and a discontinuous transmission device may be added. In that case, the problem is on the receiving side. That is, the reproduced sound is intermittent on the receiving side, resulting in a very unpleasant sound. The reason for this is that background noise is superimposed on the voice when transmitting the voice, but when the voice is not transmitted, the background noise is not transmitted, and the background noise is modulated and transmitted only when the voice signal is continuous. It is known that this is because.

【0004】このような問題点を解決する方法として、
受信側で、音声信号が伝送されてこない間は送信側の背
景雑音に類似した擬似雑音を発声させる方法が知られて
いる。図3は、上記の一例を説明する従来の音声復号装
置のブロック図である。図3において、1はアンテナ、
2は受信復調器であり、送信側符号化器の音声検出器で
検出された音声の有り無しを示す音声検出フラグとAD
PCM符号化データとが多重され変調された変調波を受
信復調する。3は受信復調された信号aをADPCM符
号化データcと音声検出フラグbとに分離する多重分離
器である。5は制御器であり、多重分離器3からの音声
検出フラグbを受け、制御信号(リセットパルス)d
と、有音/無音を示す制御信号eを出力する。
As a method of solving such a problem,
It is known that the receiving side produces pseudo noise similar to the background noise on the transmitting side while no voice signal is being transmitted. FIG. 3 is a block diagram of a conventional speech decoding apparatus for explaining the above example. In FIG. 3, 1 is an antenna,
Reference numeral 2 denotes a reception demodulator, which includes a voice detection flag indicating whether or not voice is detected by the voice detector of the transmission side encoder and AD.
A modulated wave that is modulated by being multiplexed with PCM encoded data is received and demodulated. A demultiplexer 3 demultiplexes the received and demodulated signal a into ADPCM encoded data c and a voice detection flag b. Reference numeral 5 denotes a controller, which receives a voice detection flag b from the demultiplexer 3 and receives a control signal (reset pulse) d.
And a control signal e indicating sound / silence is output.

【0005】4はADPCM復号器であり、制御器5か
らの制御信号dによって予測係数が“0”にリセットさ
れ、有音区間は多重分離器3からのADPCM符号化デ
ータcを復号し、無音区間は、擬似背景雑音用に内部発
生させたランダムADPCM符号化データを、予測係数
保持器7から入力される予測係数によって復号する。7
は予測係数保持器であり、ADPCM復号器4の内部変
数である予測係数を抽出し、フレーム毎に平均値を計算
して記憶更新するとともに、制御器5からの制御信号e
により無音区間はその直前の予測係数を保持しながらA
DPCM復号器4に与える。8はスピーカである。
Reference numeral 4 denotes an ADPCM decoder, the prediction coefficient of which is reset to "0" by the control signal d from the controller 5, and the ADPCM coded data c from the demultiplexer 3 is decoded in the voiced section to produce silence. In the section, the random ADPCM coded data internally generated for pseudo background noise is decoded by the prediction coefficient input from the prediction coefficient holder 7. 7
Is a prediction coefficient holder, which extracts a prediction coefficient which is an internal variable of the ADPCM decoder 4, calculates an average value for each frame, stores and updates it, and outputs a control signal e from the controller 5.
Therefore, in the silent section, the prediction coefficient immediately before that is retained while A
It is given to the DPCM decoder 4. Reference numeral 8 is a speaker.

【0006】図4は送信側の源音と音声復号装置の各部
の信号波形を示すタイムチャートである。図の最上段は
送信側の源音の波形であり、音声と背景雑音が重畳され
ている。2段目は送信側から送られ多重分離器3で分離
した音声検出フラグbであり、有音区間は“1”、無音
区間は“0”である。この例では、無音区間中に誤って
短時間有音を示している。3段目は制御器5から出力さ
れるリセット信号dを示す。4段目のxは、源音(音声
+背景雑音)に対する予測係数のうち2次の予測係数a
l(t)の時間変化を示す波形である。次の段のjは、
ADPCM復号器4の予測係数al(t)の時間変化を
示す波形であり、有音区間はxと同じであるが、無音区
間は予測係数保持器7から入力され擬似背景雑音を復号
する予測係数の時間変化を示す。
FIG. 4 is a time chart showing the source sound on the transmitting side and the signal waveform of each part of the speech decoding apparatus. At the top of the figure is the waveform of the source sound on the transmission side, in which voice and background noise are superimposed. The second stage is a voice detection flag b sent from the transmitting side and separated by the demultiplexer 3, in which the voiced section is "1" and the silent section is "0". In this example, the voice is erroneously shown for a short period of time during the silent section. The third stage shows the reset signal d output from the controller 5. The x in the fourth row is a quadratic prediction coefficient a among the prediction coefficients for the source sound (voice + background noise).
It is a waveform which shows the time change of l (t). The next j is
It is a waveform showing the temporal change of the prediction coefficient al (t) of the ADPCM decoder 4, and the voiced section is the same as x, but the silent section is input from the prediction coefficient holder 7 and is a prediction coefficient for decoding the pseudo background noise. Shows the change over time.

【0007】図3と図4を用いて従来の装置の動作を説
明する。音声検出フラグとADPCM符号化データが多
重化された変調波をアンテナ1で受け、受信復調器2で
受信復調して多重分離器3に復調信号aを送る。ここで
いう音声検出フラグとは送信側符号化器の入力音声の音
声が有る部分(有音区間)と無い部分(無音区間)を音
声検出器で検出した結果を示す信号である。多重分離器
3では、音声検出フラグbとADPCM符号化データc
に分離する。この場合ADPCM符号化データcは5ms
ecを1フレームとしている。5msec毎に音声検出フラグ
bが制御器5に送られる。
The operation of the conventional device will be described with reference to FIGS. 3 and 4. The modulated wave in which the voice detection flag and the ADPCM coded data are multiplexed is received by the antenna 1, received and demodulated by the reception demodulator 2, and the demodulated signal a is sent to the demultiplexer 3. The voice detection flag mentioned here is a signal indicating the result of detection by the voice detector of a portion (voiced section) of the input voice of the transmission side encoder and a portion (voiceless section) of the input voice. In the demultiplexer 3, the voice detection flag b and the ADPCM encoded data c
To separate. In this case, the ADPCM coded data c is 5 ms
ec is one frame. The voice detection flag b is sent to the controller 5 every 5 msec.

【0008】制御器5では、音声検出フラグbを受け
て、有音区間から無音区間に移行するときと、無音区間
から有音区間に移行するときにADPCM復号器4に対
して制御信号(リセットパルス)dを出力する。この制
御信号dは、ADPCM復号器4内の予測係数等の決め
られた変数を初期化するためのもので、ここでは音声検
出による送信停止時及び、送信開始時の送信側符号化器
の内部の状態と、受信側復号器の内部の状態を同じにす
るために各移行時にリセットをかける。このリセットが
ないと、音声検出による送信断後、符号化器と復号器と
の内部状態が異なり、復号器で復号した再生音声の音質
を劣化させてしまう。制御信号dによりADPCM復号
器4にリセットがかかる。また、ADPCM復号器4は
無音区間では変調信号が途絶えてしまうため、自ら擬似
背景雑音を生成するためにADPCM符号化データcが
取り得る範囲のランダムなデータを内部生成してそのデ
ータを予測係数保持器7から入力される予測係数によっ
て復号する。
Upon receiving the voice detection flag b, the controller 5 sends a control signal (reset) to the ADPCM decoder 4 at the time of transition from the voiced section to the voiceless section and at the transition from the voiceless section to the voiced section. Pulse) d is output. This control signal d is for initializing a predetermined variable such as a prediction coefficient in the ADPCM decoder 4, and here, the transmission side encoder at the time of transmission stop due to voice detection and at the time of transmission start In order to make the state of 1 and the state inside the receiving side decoder the same, a reset is applied at each transition. Without this reset, the internal states of the encoder and the decoder are different after transmission interruption due to voice detection, and the sound quality of the reproduced voice decoded by the decoder is deteriorated. The control signal d resets the ADPCM decoder 4. Further, the ADPCM decoder 4 internally generates random data within a range that the ADPCM coded data c can take in order to generate pseudo background noise by itself because the modulated signal is cut off in the silent section, and the data is used as a prediction coefficient. Decoding is performed using the prediction coefficient input from the holder 7.

【0009】予測係数保持器7は、送信断時のADPC
M復号器4で生成した擬似背景雑音に実際の雑音のスペ
クトラム情報を付加するために、ADPCM復号器4内
の予測係数を抽出してフレーム毎に平均値を計算して更
新保持し、制御信号eにより無音区間を検知したとき有
音区間の最後のフレームの予測係数の平均値を保持して
擬似背景雑音に与える。この効果でランダムなデータを
用いて復号しても実際の符号化器側の音声に重畳されて
いる背景雑音と似た音色を持つ擬似背景雑音が生成され
る。
The prediction coefficient holder 7 is used for ADPC when transmission is interrupted.
In order to add the spectrum information of the actual noise to the pseudo background noise generated by the M decoder 4, the prediction coefficient in the ADPCM decoder 4 is extracted, the average value is calculated for each frame, updated and held, and the control signal When a silent section is detected by e, the average value of the prediction coefficients of the last frame of the sound section is held and given to the pseudo background noise. With this effect, even if decoding is performed using random data, pseudo background noise having a tone color similar to the background noise superimposed on the actual speech on the encoder side is generated.

【0010】[0010]

【発明が解決しようとする課題】前述のように、図4の
xは、源音(音声+背景雑音)に対する予測係数のう
ち、例えば2次の予測係数a1(t)の時間変化を示
す。音声検出フラグが音声検出器の判定誤りや、伝送路
上でのフェージング等の影響により誤り、図4の音声検
出フラグbのように無声区間中のB区間が有声区間とし
て処理された場合、図4のjのA区間の最後でリセット
がかかる。このリセットにより“0”になった予測係数
はB区間で背景雑音に対する値に立ち上がっていく。こ
こでB区間が極めて短い区間(例えば5フレーム)のの
ち再び無音区間になると、次のC区間(無音区間)で、
予測係数が立ち上がる前の値を保持して擬似背景雑音を
作るため、区間Cでは実際の背景雑音とは異なった音色
の雑音になっしまい、受話者が違和感を感じてしまう。
すなわち図4の区間Aと区間Cの擬似背景雑音が異なっ
た音質になってしまい耳障りであるという欠点がある。
As described above, x in FIG. 4 represents the temporal change of the second-order prediction coefficient a1 (t) of the prediction coefficients for the source sound (voice + background noise). When the voice detection flag is erroneous due to the determination error of the voice detector or the influence of fading on the transmission path, and the B section in the unvoiced section is processed as the voiced section like the voice detection flag b in FIG. Reset is applied at the end of section A of j. The prediction coefficient which has become “0” by this reset rises to a value for background noise in the B section. Here, if the B section becomes a silent section after an extremely short section (for example, 5 frames), in the next C section (silent section),
Since the pseudo background noise is generated by holding the value before the prediction coefficient rises, the noise becomes different in tone color from the actual background noise in the section C, and the listener feels uncomfortable.
That is, there is a drawback that the pseudo background noises in the section A and the section C in FIG.

【0011】本発明の目的は、ADPCM音声符号化方
式における従来技術の問題点である受信側の再生音声信
号に挿入される擬似背景雑音の不自然さを解消した音声
復号装置を提供することにある。
It is an object of the present invention to provide a speech decoding apparatus which eliminates the unnaturalness of the pseudo background noise inserted in the reproduced speech signal on the receiving side, which is a problem of the prior art in the ADPCM speech coding system. is there.

【0012】[0012]

【課題を解決するための手段】本発明の音声復号装置
は、適応差分PCM符号化された符号化音声信号と音声
の有音区間と無音区間のいずれかを示す音声検出フラグ
とが多重変調され有声区間のみ送信された信号を受信し
て復調する受信復調器と、該受信復調器からの復調信号
を前記符号化音声信号と前記音声検出フラグとに分離す
る多重分離器と、該多重分離器からの前記音声検出フラ
グをそのまま出力するとともに音声検出フラグが有音区
間から無音区間,無音区間から有音区間に変わる毎にリ
セット信号を出力する制御器と、有音区間中は前記リセ
ット信号によりリセットされた0から立ち上がる予測係
数によって前記多重分離器からの符号化音声信号を復号
して再生音声信号を出力し、無音区間中はその無音区間
の直前の有音区間の背景雑音の予測係数によって内部で
発生させたランダム符号を復号して擬似背景雑音を出力
する適応差分PCM復号器と、前記制御器からの音声検
出フラグが入力され、有音区間中は前記適応差分PCM
復号器から前記予測係数を取り込んでフレーム毎に平均
値を算出して更新記憶し、無音区間中は予測係数平均値
の更新を中止し記憶された無音区間に入る直前の予測係
数平均値を前記背景雑音の予測係数として前記適応差分
PCM復号器に与える予測係数保持器とを備えた音声復
号装置において、受信した音声検出フラグが、音声が無
いとき誤って短時間、有音区間を示す誤りフラグであっ
たときの擬似背景雑音の音質の変化を軽減するために、
前記制御器と前記予測係数保持器との間に、前記制御器
からの音声検出フラグが有音区間継続中および無音区間
継続中はそのまま出力するとともに、無音区間から有音
区間に変わったとき、前記適応差分PCM復号器におい
て0にリセットされた予測係数の値が立ち上がって、当
該有音区間に変わる直前の無音区間の予測係数の値にほ
ぼ等しくなるまでの時間、無音区間を示す音声検出フラ
グを前記予測係数保持器に与えて予測係数の記憶更新を
中止したままにしておく補助制御器を設けたことを特徴
とするものである。
According to the speech decoding apparatus of the present invention, a coded speech signal subjected to adaptive differential PCM coding and a speech detection flag indicating one of a voiced section and a silent section of voice are multiplexed and modulated. A reception demodulator that receives and demodulates a signal transmitted only in a voiced section, a demultiplexer that demultiplexes the demodulated signal from the reception demodulator into the encoded voice signal and the voice detection flag, and the demultiplexer And a controller that outputs the voice detection flag as it is and outputs a reset signal each time the voice detection flag changes from a voiced section to a silent section or from a voiced section to a voiced section, and by the reset signal during the voiced section. The encoded voice signal from the demultiplexer is decoded by the reset prediction coefficient rising from 0, and the reproduced voice signal is output. During the silent period, the voiced segment immediately before the silent segment is output. An adaptive difference PCM decoder that decodes a random code internally generated by a prediction coefficient of the background noise and outputs pseudo background noise, and a voice detection flag from the controller are input, and the adaptive difference is applied during a sound section. PCM
The prediction coefficient is fetched from the decoder, the average value is calculated for each frame and updated and stored, and during the silent section, the update of the prediction coefficient average value is stopped and the predicted coefficient average value immediately before entering the stored silent section is stored. In a speech decoding device provided with a prediction coefficient holder which is given to the adaptive differential PCM decoder as a prediction coefficient of background noise, a received speech detection flag erroneously indicates a voiced section for a short time when there is no speech. In order to reduce the change in the sound quality of the pseudo background noise when
Between the controller and the prediction coefficient holder, the voice detection flag from the controller is output as it is during the voiced section continuation and the voiced section continuation, and when the voiced section is changed to the voiced section, In the adaptive differential PCM decoder, the time until the value of the prediction coefficient reset to 0 rises and becomes substantially equal to the value of the prediction coefficient of the silent section immediately before the change to the voiced section, the voice detection flag indicating the silent section. Is provided to the prediction coefficient holder, and an auxiliary controller for keeping the storage and updating of the prediction coefficient is provided.

【0013】[0013]

【実施例】図1は本発明の音声復号装置の構成例図であ
る。図1において、アンテナ1、受信復調器2、多重分
離器3、ADPCM復号器4、制御器5、予測係数保持
器7、スピーカ8は従来と同じである。6は、制御器5
からの制御信号を受けて予測係数保持器7の動作を制御
する補助制御器である。7は、ADPCM復号器4の内
部変数である予測係数gを抽出してフレーム毎に平均値
を計算し、制御器5、及び補助制御器6からの制御信号
fにより平均値の記憶と更新を行い、その値hをADP
CM復号器4に与える予測係数保持器である。
1 is a diagram showing an example of the configuration of a speech decoding apparatus according to the present invention. In FIG. 1, an antenna 1, a reception demodulator 2, a demultiplexer 3, an ADPCM decoder 4, a controller 5, a prediction coefficient holder 7, and a speaker 8 are the same as in the conventional case. 6 is a controller 5
It is an auxiliary controller that controls the operation of the prediction coefficient holder 7 by receiving a control signal from. Reference numeral 7 extracts the prediction coefficient g which is an internal variable of the ADPCM decoder 4, calculates an average value for each frame, and stores and updates the average value by the control signal f from the controller 5 and the auxiliary controller 6. And the value h is ADP
This is a prediction coefficient holder provided to the CM decoder 4.

【0014】次に、図1の本発明の実施例の動作につい
て説明する。補助制御器6及び予測係数保持器7の動作
以外は従来技術と同じであるため、図4を参照して従来
技術と異なるところを説明する。制御器5から音声検出
フラグによる有音,無音のいずれかを示す制御信号eを
受けた補助制御器6は、有音区間から有音区間に切り変
わったとき、リセットされた予測係数が立ち上がるまで
の間、予測係数保持器7の更新を中止させたままにして
おく保持器更新信号f、即ち無音区間を示す音声検出フ
ラグを予測係数保持器7に出力する。予測係数保持器7
は保持器更新信号fに従って更新/中止の動作を行う。
Next, the operation of the embodiment of the present invention shown in FIG. 1 will be described. Other than the operations of the auxiliary controller 6 and the prediction coefficient holder 7, it is the same as the conventional technique, and therefore the difference from the conventional technique will be described with reference to FIG. When the auxiliary controller 6 receives the control signal e indicating the presence or absence of the sound according to the voice detection flag from the controller 5, the auxiliary controller 6 switches from the sound section to the sound section until the reset prediction coefficient rises. During the period, the holder update signal f for keeping the updating of the prediction coefficient holder 7 stopped, that is, the voice detection flag indicating the silent section is output to the prediction coefficient holder 7. Prediction coefficient holder 7
Performs update / stop operation according to the retainer update signal f.

【0015】図2は本発明の要部をなす補助制御器6の
詳細ブロック図である。図において、61は制御信号e
を受けて有音区間カウンタ62の更新及びリセットをす
るカウンタ制御器である。62は有音区間をカウントす
る有音区間カウンタである。63は、有音カウンタのカ
ウント値と決められた定数の値を比較して結果を保持器
更新制御器64に出力する比較器である。64は比較器
63の比較結果を受けて保持器更新信号fを出力する保
持器更新制御器である。
FIG. 2 is a detailed block diagram of the auxiliary controller 6 which is a main part of the present invention. In the figure, 61 is a control signal e
In response, the counter controller updates and resets the voiced section counter 62. Reference numeral 62 is a voiced section counter that counts voiced sections. Reference numeral 63 is a comparator that compares the count value of the sound counter with the value of a predetermined constant and outputs the result to the holder update controller 64. Reference numeral 64 is a retainer update controller that outputs a retainer update signal f in response to the comparison result of the comparator 63.

【0016】次に、補助制御器6の動作について説明す
る。補助制御器6は、制御器5から入力される制御信号
e(音声検出フラグ)が無音区間から有音区間に変わっ
たときの短時間を除き、有音区間中および無音区間中を
示している間はその音声検出フラグeに対応した保持器
更新信号fを出力する。さらに、無音区間から有音区間
に変わったとき、その有音区間に変わった時点からフレ
ーム数のカウントを開始し、短時間、例えば5フレーム
区間内は引き続き無音区間を示す信号を保持器更新信号
fとして出力する。従って、保持器更新信号fを受ける
予測係数保持器7では、有音区間に変わった時点から5
フレーム以内の間は予測係数の記憶更新が中止されたま
まになる。
Next, the operation of the auxiliary controller 6 will be described. The auxiliary controller 6 indicates during the voiced section and during the voiced section, except for a short time when the control signal e (voice detection flag) input from the controller 5 changes from the voiceless section to the voiced section. In the meantime, the cage update signal f corresponding to the voice detection flag e is output. Further, when the soundless section is changed to the soundable section, counting of the number of frames is started from the time when it is changed to the soundable section, and a signal indicating the silence section is continuously output for a short time, for example, within 5 frame sections, to the holder update signal. Output as f. Therefore, in the prediction coefficient holder 7 that receives the holder update signal f, 5
The memory update of the prediction coefficient remains suspended during the frame.

【0017】この引き続き無音区間を示す保持器更新信
号fを出力する時間としては、適応差分PCM復号器4
において0にリセットされた予測係数の値が立ち上がっ
て、当該有音区間に変わる直前の無音区間における背景
雑音の予測係数の値にほぼ等しくなるまでの時間が設定
される。背景雑音を符号化および復号するときの予測係
数の分布,レベルが背景雑音の音色によって異なり、一
般的には背景雑音の予測係数の値はさほど大きくないの
で、この設定時間は5フレーム(25msec)程度でよい
が、背景雑音の音色によってその予測係数の値が大きい
場合は10フレーム(50msec)にすると効果が良好で
ある。しかし、これより長い時間を設定するときには有
音区間の話頭切れの影響を考慮する必要がある。
As the time period for which the retainer update signal f indicating the silent section is subsequently output, the adaptive differential PCM decoder 4 is used.
At the time when the value of the prediction coefficient reset to 0 rises and becomes substantially equal to the value of the prediction coefficient of the background noise in the silent section immediately before changing to the sound section, the time is set. The distribution and level of the prediction coefficient when encoding and decoding the background noise differ depending on the tone color of the background noise, and generally the value of the prediction coefficient of the background noise is not so large, so this setting time is 5 frames (25 msec). The degree is good, but when the value of the prediction coefficient is large due to the timbre of the background noise, the effect is good when it is set to 10 frames (50 msec). However, when setting a time longer than this, it is necessary to consider the effect of speech break in the voiced section.

【0018】以下、図2の詳細回路について説明する。
制御信号eを受けたカウンタ制御器61は、フレーム単
位の制御信号eが有音区間を示している間はカウンタ更
新信号kを出力して有音区間カウンタ62を順次1づつ
更新させ、無音区間を示している間はカウンタリセット
信号mを出力して有音区間カウンタ62を“0”にリセ
ットする。そして有音区間カウンタ62はその都度比較
器63にカウンタ値を出力する。比較器63には、適応
差分PCM復号器4において0にリセットされた予測係
数の値が立ち上がって、当該有音区間に変わる直前の無
音区間の背景雑音の予測係数の値にほぼ等しくなるまで
の時間を示す数値が比較定数として設定される。例え
ば、5フレーム(25msec)の場合は5が設定される。
従って、比較器63は、入力されるカウンタ値が0〜5
の範囲内かまたは5を超えたかを判定し、その比較結果
を保持器更新制御器64に出力する。
The detailed circuit of FIG. 2 will be described below.
The counter controller 61 that receives the control signal e outputs the counter update signal k while the frame-based control signal e indicates the voiced section, and sequentially updates the voiced section counter 62 by one, and outputs the silent section. Is displayed, the counter reset signal m is output to reset the voiced section counter 62 to "0". Then, the sound section counter 62 outputs the counter value to the comparator 63 each time. In the comparator 63, until the value of the prediction coefficient reset to 0 in the adaptive difference PCM decoder 4 rises and becomes almost equal to the value of the prediction coefficient of the background noise in the silent section immediately before the change to the sound section. A numerical value indicating time is set as a comparison constant. For example, 5 is set for 5 frames (25 msec).
Therefore, in the comparator 63, the input counter value is 0 to 5
It is determined whether it is within the range or exceeds 5, and the comparison result is output to the holder update controller 64.

【0019】保持器更新制御器64は、上記比較結果に
従って、結果が5を超えたときは有音区間を示す保持器
更新信号fを出力して予測係数保持器7の予測係数の更
新記憶を行わせ、0〜5の範囲内のときは無音区間を示
す保持器更新信号fを出力して予測係数保持器7の予測
係数の更新を中止させる。すなわち、予測係数保持器7
は、無音区間中および無音区間から有音区間に変わって
も5フレーム(25msec)は予測係数の保持を続ける。
即ち、図4の(f)のような保持器更新信号fが出力さ
れるので従来のjの区間Bの間は更新が中止されたまま
となり、hのように無音区間AとCの予測係数の値が連
続して同じ値になり、背景雑音は変化しないので耳障り
な雑音が解消される。
In accordance with the above comparison result, the cage update controller 64 outputs a cage update signal f indicating a voiced section when the result exceeds 5 to update and store the prediction coefficient of the prediction coefficient holder 7. If it is within the range of 0 to 5, the holder update signal f indicating the silent section is output to stop the update of the prediction coefficient of the prediction coefficient holder 7. That is, the prediction coefficient holder 7
Holds the prediction coefficient for 5 frames (25 msec) even during the silent section and when the silent section is changed to the sound section.
That is, since the retainer update signal f as shown in (f) of FIG. 4 is output, the update is stopped during the section B of the conventional j, and the prediction coefficient of the silent sections A and C is shown as h. Since the value of becomes the same value continuously, and the background noise does not change, annoying noise is eliminated.

【0020】[0020]

【発明の効果】以上詳細に説明したように、誤った有音
区間を示す音声検出フラグを受信しても、リセットによ
る予測係数の立ち上がり時間を考慮した時間、保持して
いる予測係数を擬似背景雑音に与えることにより聴感上
の不具合を解消することができる。
As described above in detail, even if a voice detection flag indicating an erroneous voiced section is received, the prediction coefficient held for a time period taking into account the rise time of the prediction coefficient due to reset is simulated. By giving noise, it is possible to solve the hearing problem.

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

【図1】本発明の音声復号器の構成例図である。FIG. 1 is a diagram showing a configuration example of a speech decoder according to the present invention.

【図2】本発明の要部をなす補助制御器の構成例図であ
る。
FIG. 2 is a diagram showing a configuration example of an auxiliary controller forming a main part of the present invention.

【図3】従来の音声復号器の構成例図である。FIG. 3 is a diagram showing a configuration example of a conventional speech decoder.

【図4】各信号の時間変化を示すタイムチャートであ
る。
FIG. 4 is a time chart showing a time change of each signal.

【符号の説明】[Explanation of symbols]

1 アンテナ 2 受信復調器 3 多重分離器 4 ADPCM復号器 5 制御器 6 補助制御器 7 予測係数保持器 8 スピーカ a 復調信号 b 音声検出フラグ c ADPCM符号化信号 d 制御信号 e 制御信号 f 保持器更新制御信号 g 予測係数 h 保持されていた予測係数 i 予測係数 k カウンタ計算信号 m カウンタリセット信号 61 カウンタ制御器 62 有音区間カウンタ 63 比較器 64 保持器更新制御器 1 antenna 2 reception demodulator 3 demultiplexer 4 ADPCM decoder 5 controller 6 auxiliary controller 7 prediction coefficient holder 8 speaker a demodulation signal b voice detection flag c ADPCM coded signal d control signal e control signal f holder update Control signal g prediction coefficient h held prediction coefficient i prediction coefficient k counter calculation signal m counter reset signal 61 counter controller 62 voiced section counter 63 comparator 64 holder update controller

Claims (1)

【特許請求の範囲】[Claims] 【請求項1】 適応差分PCM符号化された符号化音声
信号と音声の有音区間と無音区間のいずれかを示す音声
検出フラグとが多重変調され有声区間のみ送信された信
号を受信して復調する受信復調器と、該受信復調器から
の復調信号を前記符号化音声信号と前記音声検出フラグ
とに分離する多重分離器と、該多重分離器からの前記音
声検出フラグをそのまま出力するとともに音声検出フラ
グが有音区間から無音区間,無音区間から有音区間に変
わる毎にリセット信号を出力する制御器と、有音区間中
は前記リセット信号によりリセットされた0から立ち上
がる予測係数によって前記多重分離器からの符号化音声
信号を復号して再生音声信号を出力し、無音区間中はそ
の無音区間の直前の有音区間の背景雑音の予測係数によ
って内部で発生させたランダム符号を復号して擬似背景
雑音を出力する適応差分PCM復号器と、前記制御器か
らの音声検出フラグが入力され、有音区間中は前記適応
差分PCM復号器から前記予測係数を取り込んでフレー
ム毎に平均値を算出して更新記憶し、無音区間中は予測
係数平均値の更新を中止し記憶された無音区間に入る直
前の予測係数平均値を前記背景雑音の予測係数として前
記適応差分PCM復号器に与える予測係数保持器とを備
えた音声復号装置において、 受信した音声検出フラグが、音声が無いとき誤って短時
間、有音区間を示す誤りフラグであったときの擬似背景
雑音の音質の変化を軽減するために、 前記制御器と前記予測係数保持器との間に、前記制御器
からの音声検出フラグが有音区間継続中および無音区間
継続中はそのまま出力するとともに、無音区間から有音
区間に変わったとき、前記適応差分PCM復号器におい
て0にリセットされた予測係数の値が立ち上がって、当
該有音区間に変わる直前の無音区間の予測係数の値にほ
ぼ等しくなるまでの時間、無音区間を示す音声検出フラ
グを前記予測係数保持器に与えて予測係数の記憶更新を
中止したままにしておく補助制御器を設けたことを特徴
とする音声復号装置。
1. A received signal is demodulated by receiving a signal in which a coded voice signal subjected to adaptive differential PCM coding and a voice detection flag indicating any one of a voiced section and a voiceless section of voice are multiplexed and transmitted only in a voiced section. A receiving demodulator, a demultiplexer that separates the demodulated signal from the receiving demodulator into the encoded voice signal and the voice detection flag, and the voice detection flag from the demultiplexer that is output as it is A controller that outputs a reset signal each time the detection flag changes from a voiced section to a silence section, and from a silence section to a voiced section, and the demultiplexing by a prediction coefficient rising from 0 reset by the reset signal during the voiced section. The encoded voice signal from the device is decoded and the reproduced voice signal is output.In the silent period, it is generated internally by the prediction coefficient of the background noise in the voiced period immediately before the silent period. An adaptive differential PCM decoder that decodes a random code and outputs pseudo background noise, and a voice detection flag from the controller are input, and the prediction coefficient is fetched from the adaptive differential PCM decoder during a sound period. The average value is calculated and updated and stored for each frame, the update of the prediction coefficient average value is stopped during a silent section, and the average value of the prediction coefficient immediately before entering the stored silent section is used as the prediction coefficient of the background noise. In a voice decoding device provided with a prediction coefficient holder for giving to a PCM decoder, a pseudo background noise when a received voice detection flag is an error flag indicating a voiced section by mistake for a short time when there is no voice In order to reduce the change in sound quality, the voice detection flag from the controller is output as it is between the controller and the prediction coefficient holder during the voiced section and the silent section. At the same time, when the silent section is changed to the voiced section, the value of the prediction coefficient reset to 0 in the adaptive difference PCM decoder rises to almost the same value as the prediction coefficient of the silent section immediately before the change to the voiced section. A speech decoding apparatus, characterized in that an auxiliary controller is provided for giving a speech detection flag indicating a silent section to the prediction coefficient holder for keeping the storage and updating of the prediction coefficient stopped until the time becomes equal.
JP17388495A 1994-06-20 1995-06-19 Voice decoding device Expired - Fee Related JP3593183B2 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP17388495A JP3593183B2 (en) 1994-06-20 1995-06-19 Voice decoding device

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
JP6-159694 1994-06-20
JP15969494 1994-06-20
JP17388495A JP3593183B2 (en) 1994-06-20 1995-06-19 Voice decoding device

Publications (2)

Publication Number Publication Date
JPH0870285A true JPH0870285A (en) 1996-03-12
JP3593183B2 JP3593183B2 (en) 2004-11-24

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ID=26486411

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Application Number Title Priority Date Filing Date
JP17388495A Expired - Fee Related JP3593183B2 (en) 1994-06-20 1995-06-19 Voice decoding device

Country Status (1)

Country Link
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Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2000046789A1 (en) * 1999-02-05 2000-08-10 Fujitsu Limited Sound presence detector and sound presence/absence detecting method
JP2014504094A (en) * 2010-12-14 2014-02-13 フラウンホッファー−ゲゼルシャフト ツァ フェルダールング デァ アンゲヴァンテン フォアシュンク エー.ファオ Encoder and predictive encoding method, decoder and decoding method, predictive encoding and decoding system and method, and predictively encoded information signal

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2000046789A1 (en) * 1999-02-05 2000-08-10 Fujitsu Limited Sound presence detector and sound presence/absence detecting method
JP2014504094A (en) * 2010-12-14 2014-02-13 フラウンホッファー−ゲゼルシャフト ツァ フェルダールング デァ アンゲヴァンテン フォアシュンク エー.ファオ Encoder and predictive encoding method, decoder and decoding method, predictive encoding and decoding system and method, and predictively encoded information signal

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