公开(公告)号：US20080155001A1

公开(公告)日：2008-06-26

申请号：US11942654

申请日：2007-11-19

申请人： Wenshun Tian

发明人： Wenshun Tian

IPC分类号： G06F17/10

CPC分类号： H03H17/0219 ,  H03H17/0213 ,  H03H17/06

摘要： A method for long impulse response digital filtering of an input data stream, by use of a digital filtering system. Where the input data stream is divided into zero-input signals and zero-state signals. One of the zero-input signals and a corresponding impulse response of the digital filtering system is converted lo the frequency domain to determine a respective zero-input response of the digital filtering system. One of the zero-state signals is convolved with a corresponding impulse response of the digital filtering system to determine a respective zero-state response of the digital filtering system, wherein at least part of the zero-input signal precedes the zero-stale signal. The zero-state response of the digital filtering system is added to the zero-input response or the digital filtering system to determine the response of the digital filtering system. Apparatus for effecting this method is also disclosed.

摘要翻译： 一种通过使用数字滤波系统对输入数据流进行长脉冲响应数字滤波的方法。 其中输入数据流被分成零输入信号和零状态信号。 零输入信号之一和数字滤波系统的相应脉冲响应被转换成频域，以确定数字滤波系统的相应的零输入响应。 其中一个零状态信号与数字滤波系统的对应的脉冲响应进行卷积，以确定数字滤波系统的相应的零状态响应，其中零输入信号的至少一部分在零级信号之前。 数字滤波系统的零状态响应被添加到零输入响应或数字滤波系统以确定数字滤波系统的响应。 还公开了用于实现该方法的装置。

公开(公告)号：US06424687B1

公开(公告)日：2002-07-23

申请号：US09268539

申请日：1999-03-15

申请人： Wenshun Tian ,  Kah Yong

发明人： Wenshun Tian ,  Kah Yong

IPC分类号： H04L700

CPC分类号： H03H17/02 ,  H04L7/0331

摘要： A method and device to synchronize sampled digital data transferred from an input section to an output section prevents data overrun or underrun due to timing differences of timing signals of the input and output section. The timing synchronization device has an input sampled data counter to determine a number of samples in a frame time of the input sampled data. The timing synchronization device further has an interpolator to estimate data sample values for each sample of the input sampled data to coincide with each sample of the output sampled data if the number of samples in said input sampled data is less than an expected number of samples in said output sampled data. If the number of samples in said input sampled data is greater than the expected number of samples in the output sampled data, the timing synchronization device has a decimator to remove any excess samples of the input sampled data and to extrapolate each data sample of the input sampled data to coincide with each sample of the output sampled data. The timing synchronization device has a low pass filter connected to the interpolator and the decimator to prevent any aliasing of the output sampled data and a calculate and control means connected to the input sampled data counter, the interpolator, the decimator, and the low pass filter to control the operation.

摘要翻译： 将从输入部分传送的采样数字数据同步到输出部分的方法和装置防止由于输入和输出部分的定时信号的定时差异导致的数据超载或欠载。 定时同步装置具有输入采样数据计数器，用于确定输入采样数据的帧时间中的采样数。 定时同步装置还具有内插器，用于如果所述输入采样数据中的采样数量小于预期的采样数量，则估计输入采样数据的每个样本的数据采样值与输出采样数据的每个样本一致 所述输出采样数据。 如果所述输入采样数据中的样本数量大于输出采样数据中的预期采样数量，则定时同步装置具有抽取器以去除输入采样数据的任何多余样本，并且外推输入采样数据的每个数据样本 采样数据与输出采样数据的每个样本一致。 定时同步装置具有连接到内插器和抽取器的低通滤波器，以防止输出采样数据的任何混叠，以及连接到输入采样数据计数器，内插器，抽取器和低通滤波器的计算和控制装置 控制操作。

公开(公告)号：US06766020B1

公开(公告)日：2004-07-20

申请号：US09792826

申请日：2001-02-23

申请人： Wenshun Tian ,  Youhong Lu

发明人： Wenshun Tian ,  Youhong Lu

IPC分类号： H04M100

CPC分类号： G10L19/012 ,  H04B3/23

摘要： A system for creating a comfort noise signal, said system includes a clipping circuit, a comfort noise generator, and a switch. The clipping circuit receives a near-end signal, a far-end signal and a residual signal. The clipping circuit determines the power of the far-end signal, the power of the near-end signal and the power of the residual signal. The clipping circuit determines the ratio of the power of the near-end signal to the residual signal and whether the ratio is less than a predetermined threshold. A comfort noise determination block is coupled to the residual signal and selectively forms a comfort noise signal. A switch is coupled to the comfort noise determination block and the clipping circuit. The switch is activated to transmit the comfort noise signal whenever the ratio of the power of the near-end signal to the residual signal is less than a predetermined threshold.

摘要翻译： 一种用于产生舒适噪声信号的系统，所述系统包括限幅电路，舒适噪音发生器和开关。 限幅电路接收近端信号，远端信号和残留信号。 限幅电路确定远端信号的功率，近端信号的功率和剩余信号的功率。 削波电路确定近端信号的功率与剩余信号的比率以及该比率是否小于预定阈值。 舒适噪声确定块耦合到残余信号并且选择性地形成舒适噪声信号。 开关耦合到舒适噪声确定块和限幅电路。 每当近端信号的功率与残余信号的比率小于预定阈值时，开关被激活以传送舒适噪声信号。

公开(公告)号：US06590972B1

公开(公告)日：2003-07-08

申请号：US09809559

申请日：2001-03-15

申请人： Youhong Lu ,  Wenshun Tian

发明人： Youhong Lu ,  Wenshun Tian

IPC分类号： H04M100

CPC分类号： H04Q1/4575

摘要： A system and method for DTMF tone detection receives an input signal. The input signal includes a DTMF tone. The DTMF tone has a first frequency and a second frequency. An initial first frequency and an initial second frequency of said DTMF tone are selected by calculating a plurality of cost functions. The initial first frequency is confirmed to be the first frequency and the initial second frequency is confirmed to be the second frequency using re-computed values of the plurality of cost functions.

摘要翻译： 用于DTMF音检测的系统和方法接收输入信号。 输入信号包括DTMF音。 DTMF音具有第一频率和第二频率。 通过计算多个成本函数来选择所述DTMF音的初始第一频率和初始第二频率。 初始第一频率被确认为第一频率，并且使用多个成本函数的重新计算的值确认初始第二频率是第二频率。

公开(公告)号：US08340285B2

公开(公告)日：2012-12-25

申请号：US11942654

申请日：2007-11-19

申请人： Wenshun Tian

发明人： Wenshun Tian

IPC分类号： G06F17/10 ,  H04K1/04

CPC分类号： H03H17/0219 ,  H03H17/0213 ,  H03H17/06

摘要： A method for long impulse response digital filtering of an input data stream, by use of a digital filtering system. Where the input data stream is divided into zero-input signals and zero-state signals. One of the zero-input signals and a corresponding impulse response of the digital filtering system is converted to the frequency domain to determine a respective zero-input response of the digital filtering system. One of the zero-state signals is convolved with a corresponding impulse response of the digital filtering system to determine a respective zero-state response of the digital filtering system, wherein at least part of the zero-input signal precedes the zero-state signal. The zero-state response of the digital filtering system is added to the zero-input response of the digital filtering system to determine the response of the digital filtering system. Apparatus for effecting this method is also disclosed.

摘要翻译： 一种通过使用数字滤波系统对输入数据流进行长脉冲响应数字滤波的方法。 其中输入数据流被分成零输入信号和零状态信号。 零输入信号之一和数字滤波系统的相应脉冲响应被转换为频域以确定数字滤波系统的相应的零输入响应。 其中一个零状态信号与数字滤波系统的对应的脉冲响应进行卷积，以确定数字滤波系统的相应的零状态响应，其中零输入信号的至少一部分在零状态信号之前。 数字滤波系统的零状态响应被添加到数字滤波系统的零输入响应，以确定数字滤波系统的响应。 还公开了用于实现该方法的装置。

公开(公告)号：US06873701B1

公开(公告)日：2005-03-29

申请号：US09821409

申请日：2001-03-29

申请人： Wenshun Tian ,  Youhong Lu

发明人： Wenshun Tian ,  Youhong Lu

IPC分类号： H04M1/50 ,  H04Q1/457

CPC分类号： H04Q1/4575

摘要： A system and method for determining DTMF coefficients stores a plurality of reference templates, at least some of said reference templates representative of likelihood ratios. A plurality of LPC coefficients is received. The LPC coefficients are representative of an input signal. The system and method determines a plurality of current likelihood ratios based, at least in part, upon the LPC coefficients. The system and method also determines an initial tone based, at least in part, on the minimum of the current likelihood ratios and the minimum of the reference likelihood ratios. A plurality of LSF coefficients is received. The LSF coefficients are determined, at least in part, upon the input signal. The system and method verifies the validity of the initial tone based, at least in part, upon the LSF coefficients.

摘要翻译： 用于确定DTMF系数的系统和方法存储多个参考模板，表示似然比的至少一些所述参考模板。 接收多个LPC系数。 LPC系数代表输入信号。 该系统和方法至少部分地基于LPC系数确定多个当前似然比。 系统和方法还基于至少部分地基于当前似然比的最小值和参考似然比的最小值来确定初始音调。 接收多个LSF系数。 LSF系数至少部分地基于输入信号来确定。 该系统和方法至少部分地基于LSF系数验证初始音调的有效性。

公开(公告)号：US06738733B1

公开(公告)日：2004-05-18

申请号：US10089758

申请日：2002-07-22

申请人： Wenshun Tian

发明人： Wenshun Tian

IPC分类号： G01L500

CPC分类号： G10L19/12 ,  G10L2019/0008

摘要： A method and apparatus for reducing the computational load of a dual-rate encoding system having a multi-pulse maximum likelihood quantization process configured to transmit at a first transmission rate and to search subframes of excitation signals according to a reduced number of gain scale factors; and an algebraic code-excited linear prediction block configured to perform a first correlation threshold test for entry into an embedded signal processing loop and a second correlation threshold test for entry into a previous signal processing loop in which the embedded signal processing loop is embedded to reduce the number of times the previous signal processing loop and the embedded signal processing loop are entered, thereby reducing the computational load of the system.

摘要翻译： 一种用于降低具有多脉冲最大似然量化处理的双速率编码系统的计算负荷的方法和装置，其被配置为以第一传输速率进行发送，并根据减少的增益比例因子数来搜索激励信号的子帧; 以及代数码激励线性预测块，被配置为执行用于进入嵌入式信号处理循环的第一相关阈值测试和用于进入先前信号处理循环的第二相关阈值测试，其中嵌入所述嵌入信号处理循环以减少 进入先前信号处理循环和嵌入信号处理循环的次数，从而减少系统的计算负荷。

公开(公告)号：US06327689B1

公开(公告)日：2001-12-04

申请号：US09298449

申请日：1999-04-23

申请人： Wenshun Tian

发明人： Wenshun Tian

IPC分类号： G06F1110

CPC分类号： H04L1/0057 ,  G10L19/005 ,  H03M13/151 ,  H03M13/1515 ,  H04L1/0041 ,  H04L1/0071

摘要： In this invention digital audio data transmitted by wireless means is error corrected and concealed to remove and hide noise created errors ranging from random to burst noise. The data is interleaved into even and odd sub-frames to combat burst mode noise, and ECC is created for the MSB of the data and for command and control bytes using a Reed Solomon encoder before transmission. The LSB are not encoded for reasons of bandwidth because experiments have show the LSB have little effect on audible noise even at a bit error rate of 3.0E-3. The transmitted data is decoded using Reed Solomon decoder and error corrected. The digital audio data is then processed through a concealment procedure that hides the remaining MSB errors by using extrapolation, soft muting and muting depending on the state of audio data preceding and following the current sub-frame of the digital audio data. Soft muting is a form of windowing using Hanning or other windowing algorithms where the coefficients of the window algorithm diminish to a minimum at the boundaries of the data frames.

摘要翻译： 在本发明中，通过无线装置发送的数字音频数据被纠错和隐藏，用于去除和隐藏从随机到突发噪声的噪声产生的误差。 将数据交错成偶数和奇数子帧以对抗突发模式噪声，并且在传输之前使用Reed Solomon编码器为数据的MSB和命令和控制字节创建ECC。 由于实验显示，即使在3.0E-3的误码率下，LSB对可听噪声几乎没有影响，因此LSB不会被编码。 使用Reed Solomon解码器对发送的数据进行解码并进行纠错。 然后，通过隐藏程序来处理数字音频数据，该隐藏程序根据数字音频数据的当前子帧之前和之后的音频数据的状态，通过使用外推，软静音和静音来隐藏剩余的MSB误差。 软静音是使用汉宁或其他窗口算法的窗口形式，其中窗口算法的系数在数据帧的边界处减小到最小。

公开(公告)号：US06295520B1

公开(公告)日：2001-09-25

申请号：US09268540

申请日：1999-03-15

申请人： Wenshun Tian

发明人： Wenshun Tian

IPC分类号： G10L1912

CPC分类号： G10L19/10

摘要： Speech is synthesized by optimizing frame data containing an excitation signal and impulse response filter coefficients, and convolving the excitation signal and impulse response filter coefficients more efficiently and with fewer multiplications and additions. The method to convolve begins by determining a number of non-zero pulses within said excitation signal. The pulse locations are sorted for the zero and non-zero pulses. The non-zero pulses are then ranked in order of time. The codebook contributions for the synthesized output signal having an index value less than a lowest rank non-zero pulse are set to a zero value. Each remaining codebook contribution for the synthesized signal is determined by convolving each non-zero pulse within said excitation signal with each impulse response function.

摘要翻译： 通过优化包含激励信号和脉冲响应滤波器系数的帧数据，并且更有效地并且更少的乘法和加法来卷积激励信号和脉冲响应滤波器系数来合成语音。 通过确定所述激励信号内的非零脉冲数来开始卷积的方法。 为零和非零脉冲对脉冲位置进行排序。 然后将非零脉冲按时间顺序排列。 具有小于最低秩非零脉冲的指标值的合成输出信号的码本贡献被设置为零值。 通过用每个脉冲响应函数卷积所述激励信号内的每个非零脉冲来确定合成信号的每个剩余码本贡献。