公开(公告)号：US4097860A

公开(公告)日：1978-06-27

申请号：US766848

申请日：1977-02-09

申请人： Takashi Araseki ,  Kazuo Ochiai ,  Rikio Maruta

发明人： Takashi Araseki ,  Kazuo Ochiai ,  Rikio Maruta

IPC分类号： H03F3/34 ,  H03H17/02 ,  H03M1/00 ,  H04L25/03 ,  H04L25/06 ,  H03K13/02 ,  H03K13/32

CPC分类号： H03M1/1295

摘要： Method and apparatus for cancelling a deviation in an input signal due to drift, line or background noise, changes in circuit component characteristics due to aging, and the like wherein the input signal, typically a digitized representation of an analog signal, is altered by a presumed offset magnitude; the polarity of the difference is monitored and a negative or positive count of clock pulses is accumulated dependent upon the aforesaid polarity. When a determined positive (or negative) count is reached the presumed offset is adjusted (up or down) by a predetermined increment and the count is begun anew. As an alternative technique and embodiment the initiation of the count may be restrained as long as the magnitude of the digitized input signal exceeds a predetermined threshold.

摘要翻译： 用于消除由漂移，线路或背景噪声引起的输入信号偏差的方法和装置，由于老化引起的电路元件特性的变化等，其中输入信号（通常是模拟信号的数字化表示）被改变为 推定偏移量; 监视差异的极性，并根据上述极性累积时钟脉冲的负数或正数。 当达到确定的正（或负）计数时，推定的偏移量被调整（向上或向下）预定的增量，并且重新开始计数。

公开(公告)号：US4001505A

公开(公告)日：1977-01-04

申请号：US564644

申请日：1975-04-03

申请人： Takashi Araseki ,  Kazuo Ochiai

发明人： Takashi Araseki ,  Kazuo Ochiai

IPC分类号： H04M11/00 ,  G10L11/00 ,  G10L11/02 ,  G10L15/00 ,  G10L15/02 ,  G10L15/04 ,  H04J3/17 ,  G10L1/04

CPC分类号： G10L25/78 ,  H04J3/175

摘要： A speech detector which rapidly detects the presence of speech signals in telephone channel broadband noise is useful in TASI type systems. The incoming analog speech signal is sampled and binary-coded, then fed simultaneously to two different detectors: (1) A large-amplitude threshold detector (which detects high-energy vowel and plosive sounds): (2) a high-frequency threshold detector (which detects high-frequency fricative sounds above a 2 Khz threshold, assuming inherent noise in a 4 Khz telephone channel is mostly flat/broadband and 8 Khz sampling noise is filtered out).

公开(公告)号：US4028506A

公开(公告)日：1977-06-07

申请号：US530620

申请日：1974-12-09

申请人： Takashi Araseki ,  Kazuo Ochiai

发明人： Takashi Araseki ,  Kazuo Ochiai

IPC分类号： H04B3/20 ,  H04B3/22

CPC分类号： H04B3/20

摘要： A maximum value tracing circuit employed in echo suppressors and the like for selecting the peak values of digitized voice signals. The digitized value of a signal previously stored in a register is continuously compared with the present digitized value. The larger value of the compared signals is stored in the register after application of each clear signal to the register.First and second maximum value tracing circuits of the type described may be employed in combination with a comparator-selector circuit for comparing the outputs of the first and second maximum value extraction circuits to produce a comparison output signal representative of the larger of the two compared outputs derived from the first and second maximum value extraction circuits, which output signal is utiized for providing reliable, high speed echo suppression.

摘要翻译： 在回波抑制器等中采用的用于选择数字化语音信号的峰值的最大值跟踪电路。 预先存储在寄存器中的信号的数字化值与当前的数字化值不断进行比较。 在将每个清除信号应用到寄存器之后，比较信号的较大值被存储在寄存器中。

公开(公告)号：US4012603A

公开(公告)日：1977-03-15

申请号：US606791

申请日：1975-08-22

申请人： Takashi Araseki ,  Kazuo Ochiai

发明人： Takashi Araseki ,  Kazuo Ochiai

IPC分类号： H04B3/20

CPC分类号： H04B3/20

摘要： An echo suppressor employed in systems interconnecting a two-wire circuit with a four-wire circuit for substantially eliminating "talker's echo" and for selectively attenuating the calling party's voice signal when the called party at the two-wire line begins speaking. The signal levels of signals on both paths of the four-wire circuit are measured to extract the maximum values of the signals for succeeding time intervals. The extracted value of the outgoing path signal is divided by the extracted value of the incoming path signal to generate an attenuation factor value. This value is multiplied by the value of the voice signal in the incoming path to generate an estimated leakage level signal. When this level is greater than the signal level in the output path (i.e. when there is no outgoing voice signal) the outgoing path is disconnected to prevent a talker's echo signal from being transmitted thereby. When the signal on the outgoing path is greater than the estimated leakage level signal the outgoing path is not disconnected, and an attenuation circuit in the incoming path is activated to attenuate signals in the incoming path. Delay means are provided to prevent the disconnection of the outgoing path at the initial and terminal ends of a voice signal.BACKGROUND OF THE INVENTIONLong-distance communication networks are typically comprised of a four-wire circuit wherein the carriers or subcarriers are respectively allotted to outgoing and incoming information signals. A local toll circuit normally connects the four-wire circuit, through terminal equipment, to a calling or called subscriber set which is comprised of the two-wire circuit. A four-wire terminating set (or four-wire to two-wire conversion device) must therefore be provided at the terminal equipment to suitably connect the two-wire and four-wire circuits. The conventional four-wire terminating set is comprised of a hybrid coil and an impedance balancing network for attaining impedance balance between the two and four-wire circuits. However, inasmuch as the impedance of the two-wire circuit varies by an appreciable amount as a function of the distance between the terminal equipment and the local subscriber set at the extremity of the two-wire circuit or according to the performance of telephone exchanges and transmission lines between the terminal equipment and the subscriber set, it can not be expected, as a practical matter, to attain impedance balance for any arbitrary subscriber sets.For the convenience of explanation, it is assumed hereinafter that a calling subscriber at the remote end of the long-distance communication network calls a local subscriber on this side of the network. In the case where the above-mentioned impedance balance is attained, the information signal of the calling (remote) subscriber supplied through the communication network to the four-wire input terminal part of the above-mentioned hybrid coil passes without any ill effect through the hybrid coil and the two-wire circuit to the called (local) subscriber set. In the case where the impedance is not completely balanced, the information signal leaks through the hybrid coil to appear at the output terminal pair of the four-wire circuit thereof, and then is sent back to the calling subscriber, causing the phenomenon commonly referred to as talker's echo. Although the talker's echo produced in a comparatively short distance network does not cause much disturbance in the conversation, the echo produced in a long-distance communication network appreciably affects the quality and performance of the conversation, because the speech of the calling subscriber returns to the person speaking as the talker's echo after a transmission period of the order of several hundred milliseconds.An echo suppressor conventionally used for suppressing or removing the talker's echo has a structure adapted to compare the outgoing signal level at the four-wire output terminal pair of the hybrid coil with a threshold level related to the incoming signal level. so that the outgoing signal circuit can be selectively disconnected to interrupt the outgoing signal level when the result of comparison shows that the outgoing signal level is not higher than the incoming signal level. The threshold level is given a value smaller than the incoming signal level by a prefixed rate. Statistics show that the distribution of echo attenuation factor of hybrids (being the main component for the attenuation in the echo path) throughout the United States has an average of 15 dB with a standard deviation of 3 dB. Therefore, the prefixed rate is selected about 6 dB, thereby to ensure that the threshold level is greater than the leakage (echo) level for most of hybrids (at a 99.7 -percent probability). An example of a conventional echo suppressor of the amplitude comparison type particularly adapted to time-division multiplex PCM information signals is described in detail in the technical report by E. Fariello entitled "A Digital Echo Suppressor for Satellite Circuits" published in the IEEE Transactions on Communications, December 1972. Therefore, further description will not be given here.Generally, there are two mutually exclusive problems that one must deal with in an echo suppressor: reduction of voice clipping of the called subscriber's signal and reduction of echo of calling subscriber's signal. In the case of the conventional echo suppressor described above, the calling subscriber's signal is detected with the threshold level higher than the echo level by 9 dB on the average, so that the reduction of the echo is well attained but the malfunction attributed to the voice clipping appreciably reduces the conversation quality. Especially, the beginning part of a speech having relatively lower level than the remaining part is hardly detected until the level thereof reaches the threshold level set above the outgoing signal level in the state of no called subscriber's signal by 9 dB on the average, resulting in the interruption of the leading part of the speech called "initial clipping". Futhermore, if the attenuation in the echo path is poorer than 6 dB, the threshold level is needed to be adjusted more closely to the incoming signal level itself, bringing about a longer period of initial clipping. The initial clipping is inherent to the conventional echo suppressor of this type that detects the presence of the called subscriber's signal through the level comparison thereof with the threshold level which is not related to the attenuation factor of the echo path associated with the present echo suppressor.BRIEF DESCRIPTION OF THE INVENTION AND OBJECTSIt is therefore an object of this invention to provide an echo suppressor which makes it possible to suppress or remove the talker's echo without being affected by the initial clipping.Another object of the instant invention is to provide a novel electronic device for suppressing the talker's echo in a long-distance communication network, coupling a two-wire circuit at the end terminals of a four-wire circuit wherein means are provided for disconnecting the outgoing signal circuit of the four-wire circuit during the presence of the calling subscriber's signal, means for inhibiting the disconnection of the outgoing signal circuit in the presence of the called subscriber's signal detected by the comparison of the outgoing signal level with an estimated leakage (echo) component level.In the present invention, the threshold level for the detection of the called subscriber's outgoing signal is given by the estimated leakage (echo) level generated as follows:A level ratio between the outgoing signal level on the four-wire output terminal pair and incoming signal level on the four-wire input terminal pair is measured on a real time basis, and then multiplied with the incoming signal level to generate the product, i.e., the estimated leakage level. When the called subscriber's signal is not present, the level ratio indicates the real attenuation factor of the leakage (echo) path, and therefore the estimated leakage level is close to the real leakage level. To use the estimated leakage level as the threshold level, a certain delay is given to the attenuation-factor-representing signal, making it possible to avoid the initial clipping. In the presence of the called subscriber's signal, the measured level ratio tends to indicate greater value than the real attenuation factor. The ratio is set to be unity at most. It follows therefore that the sensitivity for the called subscriber's is equal, even in the worst condition, to that of the conventional echo suppressor, where the calling subscriber's signal level itself is used as the threshold level without any attenuation.

摘要翻译： 用于将双线电路与四线电路相互连接的系统中的回波抑制器，用于基本上消除“讲话人的回声”，并且当双线线路上的被叫方开始说话时选择性地衰减主叫方的语音信号。 测量四线电路两路信号的信号电平，以提取后续时间间隔信号的最大值。 输出路径信号的提取值除以输入路径信号的提取值，生成衰减因子值。 该值乘以输入路径中的语音信号的值，以产生估计的泄漏电平信号。 当该电平大于输出路径中的信号电平时（即，当没有输出语音信号时），输出路径被断开以防止由此发送说话人的回波信号。 当输出路径上的信号大于估计的泄漏电平信号时，输出路径不断开，并且激活输入路径中的衰减电路以衰减输入路径中的信号。 提供延迟装置以防止语音信号的起始端和终端处的输出路径的断开。

公开(公告)号：US4317208A

公开(公告)日：1982-02-23

申请号：US79724

申请日：1979-09-28

申请人： Takashi Araseki ,  Kazuo Ochiai

发明人： Takashi Araseki ,  Kazuo Ochiai

IPC分类号： H03M3/04 ,  H04B14/06 ,  H04L25/16

CPC分类号： H03M3/042

摘要： An adaptive differential pulse code modulation (ADPCM) system includes a predictor which predicts a sample value based on past prediction errors and coefficients which are adaptively corrected to lessen the difference, i.e. the prediction error, between the predicted values and the actual values. The predictor is duplicated in the receiver, has no feedback loop and thus instability due to transmission errors is eliminated. The system can also include a second predictor whose output is combined with that of the first predictor to obtain the predicted value. The second predictor output is based on past sums of the prediction error and the predicted value and coefficients which are adaptively corrected. The second predictor is in a feedback loop but instability is prevented by choosing the coefficients used.

摘要翻译： 自适应差分脉冲编码调制（ADPCM）系统包括预测器，该预测器基于过去的预测误差和自适应校正的系数预测样本值，以减小预测值和实际值之间的差异，即预测误差。 预测器在接收机中重复，没有反馈环路，因此消除了由于传输错误导致的不稳定性。 该系统还可以包括第二预测器，其输出与第一预测器的输出组合以获得预测值。 第二预测器输出是基于预测误差和自适应校正的预测值和系数的过去和。 第二个预测器处于反馈回路中，但是通过选择所使用的系数来防止不稳定性。

公开(公告)号：US4670903A

公开(公告)日：1987-06-02

申请号：US393893

申请日：1982-06-30

申请人： Takashi Araseki ,  Kazuo Ochiai

发明人： Takashi Araseki ,  Kazuo Ochiai

IPC分类号： H04B3/20 ,  H04B3/23 ,  H04M1/60 ,  H04M9/00

CPC分类号： H04B3/23 ,  H04B3/238

摘要： For use in combination with a loudspeaker and at least one microphone, for example, by attendants in an auditorium, an echo cancelling circuit comprises a self-adaptive echo canceller responsive to a lower frequency component, such as below 1.7 kHz, of a receive-in signal for self-adatively cancelling a corresponding component of a reverberation signal included in a send-in signal during each interval during which an audio signal reaches the circuit from a remote party. For a higher frequency reverberation signal component, an echo suppressor or a voice switch may reduce a weaker one of two signals which are either the higher frequency send-in and receive-in signal components or a combination of a reverberation component cancelled signal with the higher frequency send-in signal component and the whole receive-in signal. Alternatively, a less expensive echo canceller non-adaptively cancels a part of the reverberation signal in response to the receive-in signal. The lower frequency component of the partially reverberation cancelled signal is used by the self-adaptive echo canceller as the lower frequency send-in signal component. An acoustic output may once be reproduced by the loudspeaker in response to the receive-in signal and then converted to an electric signal for supply to the echo cancelling circuit.

摘要翻译： 为了与扬声器和至少一个麦克风组合使用，例如通过礼堂中的演奏者，回声消除电路包括响应于低于1.7kHz的较低频率分量的自适应回声消除器， 用于在音频信号到达来自远程方的电路的每个间隔期间自我消除包含在送入信号中的混响信号的对应分量。 对于较高频率混响信号分量，回波抑制器或语音切换可以减少两个信号中较弱的一个信号，这两个信号是较高频率的入接信号分量或接收信号分量，或混响分量消除信号的组合较高 频率送入信号分量和整个接收信号。 或者，较便宜的回波消除器响应于接收信号而自适应地抵消混响信号的一部分。 部分混响取消信号的较低频率成分由自适应回波消除器用作低频发送信号分量。 声输出可以一旦被扬声器响应于接收信号再现，然后转换成电信号供给回波消除电路。

公开(公告)号：US4167653A

公开(公告)日：1979-09-11

申请号：US895561

申请日：1978-04-12

申请人： Takashi Araseki ,  Kazuo Ochiai

发明人： Takashi Araseki ,  Kazuo Ochiai

IPC分类号： H04B3/20 ,  H04J3/17 ,  H04J6/00

CPC分类号： H04J3/175 ,  H04B3/20

摘要： An adaptive speech signal detector for use in a 4-wire telephone channel performs an adaptive threshold value setting operation depending on the channel noise level on a transmitter-side channel to detect a speech signal present at the transmitter. The adaptive operation of the speed signal detector is inhibited, however, if the signal level at the related receiver-side channel becomes higher than a preset value. This permits the use of the adaptive speech signal detector with DSI (digital speech interpolation) systems without malfunction due to the operation of an echo suppressor.

摘要翻译： 用于4线电话信道的自适应语音信号检测器根据发射机侧信道上的信道噪声电平执行自适应阈值设置操作，以检测出现在发射机处的语音信号。 然而，如果相关接收机侧信道的信号电平变得高于预设值，则禁止速度信号检测器的自适应操作。 这允许使用具有DSI（数字语音插值）系统的自适应语音信号检测器，由于回波抑制器的操作而没有故障。

公开(公告)号：US4005277A

公开(公告)日：1977-01-25

申请号：US610372

申请日：1975-09-04

申请人： Takashi Araseki ,  Kazuo Ochiai ,  Takashi Ogihara ,  Shunji Tanaka

发明人： Takashi Araseki ,  Kazuo Ochiai ,  Takashi Ogihara ,  Shunji Tanaka

IPC分类号： H04B3/23 ,  H04B3/20 ,  H04M9/10

CPC分类号： H04B3/20

摘要： An echo controller comprises a parallel circuit of an echo suppressor and a self-adaptive echo canceller near an echo path of a long-distance telephone network. The echo controller further comprises a mode switch responsive to levels of a received signal, an unprocessed signal, and an output signal of the echo canceller and operable during absence of double talk for suspending operation of the echo canceller and putting instead the echo suppressor into operation only when characteristics of the echo path and operation of the echo canceller are objectionable.

摘要翻译： 回波控制器包括在远程电话网络的回波路径附近的回波抑制器和自适应回波消除器的并联电路。 回波控制器还包括响应于接收信号，未处理信号和回波消除器的输出信号的电平的模式开关，并且在不存在双重通话期间可操作以暂停回波消除器的操作并且将回波抑制器置于正常工作状态 只有当回波路径的特性和回波消除器的操作是令人反感的。

公开(公告)号：US4354273A

公开(公告)日：1982-10-12

申请号：US156990

申请日：1980-06-09

申请人： Takashi Araseki ,  Kazunori Ozawa

发明人： Takashi Araseki ,  Kazunori Ozawa

IPC分类号： H03M3/04 ,  H04B15/00

CPC分类号： H03M3/042

摘要： In an adaptive differential pulse code modulation (ADPCM) system for frequency band compression of speech or like signals, the coefficient of the synthesis filter in both the transmitter and receiver is varied in accordance with the normalized error e.sub.j /.DELTA. rather than the error itself, thus providing greater frequency band compression and preventing transmission errors from rendering the synthesis filter unstable.

摘要翻译： 在用于语音或类似信号的频带压缩的自适应差分脉码调制（ADPCM）系统中，发射机和接收机中的合成滤波器的系数根据归一化误差ej / DELTA而不是错误本身而变化， 从而提供更大的频带压缩并防止传输错误使得合成滤波器不稳定。

公开(公告)号：US4716592A

公开(公告)日：1987-12-29

申请号：US565804

申请日：1983-12-27

申请人： Kazunori Ozawa ,  Takashi Araseki

发明人： Kazunori Ozawa ,  Takashi Araseki

IPC分类号： G10L19/00 ,  G10L19/10 ,  G10L5/00

CPC分类号： G10L19/10

摘要： A voice encoding system is constituted by a short time voice signal series producing circuit inputted with a discrete voice signal series for dividing the same at each short time; a parameter extracting circuit for extracting a parameter representative of a spectrum envelope from the short time voice signal series and encoding the parameter; an impulse response series calculating circuit for calculating the impulse response series based on the parameter representative of the spectrum envelope; an autocorrelation function sequence calculating circuit utilizing the impulse response series; a cross-correlation function sequence calculating circuit utilizing the impulse response series and the short time voice signal series; a circuit for calculating and encoding an excitation signal series of the short time voice signal series by utilizing the autocorrelation function sequence; and a circuit for combining and outputting a code of the parameter representative of the spectrum envelope and a code representative of the excitation signal series. With the system, high quality voice encoding can be made at a transmission rate of less than 10k bits/second with a relatively small amount of calculation.

摘要翻译： 语音编码系统由短时语音信号序列产生电路构成，该电路输入用于在短时间内对其分频的离散语音信号序列; 参数提取电路，用于从短时语音信号序列中提取表示频谱包络的​​参数，并对参数进行编码; 脉冲响应串行计算电路，用于基于表示频谱包络的​​参数来计算脉冲响应序列; 利用脉冲响应序列的自相关函数序列计算电路; 利用脉冲响应序列和短时语音信号序列的互相关函数序列计算电路; 用于通过利用自相关函数序列来计算和编码短时语音信号序列的激励信号序列的电路; 以及用于组合和输出表示频谱包络的​​参数的代码和表示激励信号序列的代码的电路。 利用该系统，可以以相对少的计算量以低于10k位/秒的传输速率进行高质量的语音编码。