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    • 1. 发明授权
    • Frequency domain format enhancement
    • 频域格式增强
    • US07430507B2
    • 2008-09-30
    • US11513804
    • 2006-08-31
    • Richard L. Zinser, Jr.Steven R. Koch
    • Richard L. Zinser, Jr.Steven R. Koch
    • G10L21/00
    • H04W88/181G10L19/00G10L19/0204G10L19/06G10L19/173G10L25/78
    • A method for enhancing speech prediction coefficient signals, including: a) receiving a set of prediction coefficient signals; b) calculating a set of amplitude spectrum coefficient signals based on the prediction coefficient signals; c) calculating a set of enhancement coefficient signals based on the amplitude spectrum coefficient signals; d) calculating a set of enhanced autocorrelation coefficient signals based on the enhancement coefficient signals; e) converting at least some of the enhanced autocorrelation coefficient signals into reflection coefficient signals; f) testing the range of values of the reflection coefficient signals; and g) converting at least some of the tested reflection coefficient signals into prediction coefficient signals.
    • 一种用于增强语音预测系数信号的方法,包括:a)接收一组预测系数信号; b)基于预测系数信号计算一组振幅谱系数信号; c)基于幅度谱系数信号计算一组增强系数信号; d)基于增强系数信号计算一组增强自相关系数信号; e)将所述增强自相关系数信号中的至少一些转换为反射系数信号; f)测试反射系数信号的值的范围; 以及g)将所测试的反射系数信号中的至少一些转换为预测系数信号。
    • 2. 发明授权
    • Two-input crosstalk-resistant adaptive noise canceller
    • 双输入抗串扰自适应噪声消除器
    • US4649505A
    • 1987-03-10
    • US627251
    • 1984-07-02
    • Richard L. Zinser, Jr.Seth D. SilversteinSteven R. Koch
    • Richard L. Zinser, Jr.Seth D. SilversteinSteven R. Koch
    • G10K11/178G10L21/02H03H21/00H03K5/159G06F7/38H04B3/20
    • H03H21/00G10K11/1784G10K2210/108G10K2210/3012G10K2210/505G10L2021/02165G10L21/0208
    • A two-input crosstalk-resistant adaptive noise canceller receives a primary input signal including a desired speech signal portion and an undesired noise signal portion and also receives a reference input signal having a reference noise input portion and a crosstalk speech portion. The canceller has first and second summer means and first and second adaptive filter means. The first summer means provides a canceller output signal which is the difference between the primary input signal and the first adaptive filter output signal. The canceller output signal is applied to the reference input of the second adaptive filter and to one of a pair of error-control inputs of the first adaptive filter. The second error-control input of the first adaptive filter is provided by the signal at the output of the second adaptive filter, which receives a single error-control input signal from the output of the second summer means. The second summer provides an output signal which is the difference between the reference input signal and the second adapter filter output signal. With the correlation bias between the desired primary input (speech) signal and the crosstalk (speech) signal in the reference input substantially reduced, the canceller output signal is then related substantially only to the primary input desired signal.
    • 双输入抗串扰自适应噪声消除器接收包括期望的语音信号部分和不期望的噪声信号部分的主要输入信号,并且还接收具有参考噪声输入部分和串扰语音部分的参考输入信号。 消除器具有第一和第二夏季装置以及第一和第二自适应滤波器装置。 第一个夏季装置提供一个消除器输出信号,它是主要输入信号和第一个自适应滤波器输出信号之间的差值。 消除器输出信号被施加到第二自适应滤波器的参考输入和第一自适应滤波器的一对误差控制输入中的一个。 第一自适应滤波器的第二误差控制输入由第二自适应滤波器的输出处的信号提供,第二自适应滤波器的输出从第二加法装置的输出接收单个误差控制输入信号。 第二个夏天提供输出信号,它是参考输入信号和第二适配器滤波器输出信号之间的差值。 由于所需的主输入(语音)信号和参考输入中的串扰(语音)信号之间的相关偏差大大降低,所以消除器输出信号然后基本上仅与主要输入的期望信号相关。
    • 6. 发明授权
    • TDVC-to-MELP transcoder
    • TDVC-MELP代码转换器
    • US06678654B2
    • 2004-01-13
    • US09991695
    • 2001-11-26
    • Richard L. Zinser, Jr.Steven R. Koch
    • Richard L. Zinser, Jr.Steven R. Koch
    • G10L1900
    • H04W88/181G10L19/00G10L19/0204G10L19/06G10L19/173G10L25/78
    • The system and method of the present invention comprises a compressed domain universal transcoder that transcodes a bit stream representing frames of data encoded according to a first compression standard (TDVC coding standard) to a bit stream representing frames of data according to a second compression standard (MELP coding standard). The method includes decoding a bit stream into a first set of parameters compatible with a first compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard without converting the first set of parameters to an analog or digital waveform representation. Lastly, the second set of parameters are encoded into a bit stream compatible with the second compression standard.
    • 本发明的系统和方法包括压缩域通用代码转换器,其将表示根据第一压缩标准(TDVC编码标准)编码的数据的比特流转码为表示根据第二压缩标准的数据帧的比特流( MELP编码标准)。 该方法包括将比特流解码成与第一压缩标准兼容的第一组参数。 接下来,将第一组参数变换为与第二压缩标准兼容的第二组参数,而不将第一组参数转换为模拟或数字波形表示。 最后,第二组参数被编码成与第二压缩标准兼容的比特流。
    • 7. 发明授权
    • Integrated data link concept for air traffic control applications
    • 空中交通管制应用的综合数据链路概念
    • US5440544A
    • 1995-08-08
    • US173226
    • 1993-12-27
    • Richard L. Zinser, Jr.
    • Richard L. Zinser, Jr.
    • H04B7/185H04J4/00H04M11/00
    • H04J4/00H04B7/18502
    • An integrated full duplex radio data link for aircraft capable of a sustained 4 to 8 Mb/sec raw data transfer rate communicates on newly proposed dedicated digital channels (DDC) for radio communications between the existing analog VHF amplitude modulated (AM) voice channels, and causes no perceptible interference to the remaining AM voice channels. While the present invention is intended primarily for terminal areas and airport surface applications, it may be employed with enroute applications. The present invention transmits AM encoded digital aircraft identification information appended to AM voice transmissions. An ground unit strips off the aircraft identification and assigns a DDC frequency to that aircraft unit over a control uplink channel (CUC) for further data communication. Downlink information, such as acknowledgement of the DDC assigned is transmitted over a digital control downlink channel (CDC). Data communication then proceeds on the DDC. The data is transmitted as a low power encoded signal having a constant envelope, and narrow bandwidth. Any interference from the DDC into the AM voice channel appears as background "white noise" to the AM receiver. The channel assignment scheme allows the present invention to function at an acceptable bit error rate.
    • 用于能够持续4至8Mb /秒原始数据传输速率的飞行器的集成全双工无线电数据链路在新提出的专用数字信道(DDC)上通信,用于现有模拟VHF调幅(AM)语音信道之间的无线电通信,以及 对剩余的AM语音信道不产生可察觉的干扰。 虽然本发明主要用于终端区域和机场地面应用,但是它可以用于路由应用。 本发明发送附加到AM语音传输的AM编码数字飞行器识别信息。 地面单元剥离飞行器识别,并通过控制上行链路信道(CUC)向该飞机单元分配DDC频率,用于进一步的数据通信。 通过数字控制下行链路信道(CDC)发送诸如分配的DDC的确认的下行链路信息。 数据通信随后在DDC上进行。 数据作为具有恒定包络和窄带宽的低功率编码信号发送。 任何来自DDC到AM语音通道的干扰都将作为AM接收机的背景“白噪声”出现。 信道分配方案允许本发明以可接受的比特错误率工作。
    • 8. 发明授权
    • Method and apparatus for generating fill frames for voice over internet protocol (VoIP) applications
    • 用于产生用于网际语音协议(VoIP)应用的填充帧的方法和装置
    • US07738361B2
    • 2010-06-15
    • US11940591
    • 2007-11-15
    • Richard L. Zinser, Jr.Martin W. Egan
    • Richard L. Zinser, Jr.Martin W. Egan
    • G06F11/00
    • H04L65/607G10L19/005H04L65/1059H04L65/80
    • A method and apparatus that generates fill frames for Voice over Internet Protocol (VoIP) applications in a communication device is disclosed. The method may include determining if there is a lost frame in a received communication, wherein if it is determined that there is a lost frame, setting a frame loss flag and storing the frame loss flag in the frame loss history register, shifting a loss history register, a line spectral frequency (LSF) history register, a voicing cutoff (VCUT) history register, a pitch history register, and a root mean squared (RMS) gain history register, wherein the loss history register, the LSF history register, the VCUT history register, the pitch history register, and the RMS history register include at least three registers, the three registers being a newest, a middle and an oldest registers, reading the frame loss flag into a newest loss history register, determining contents of the middle register of each of the LSF history register, the VCUT history register, the pitch history register, and the RMS history register, and sending the contents of the middle registers to a synthesizer to generate an output speech signal.
    • 公开了一种在通信设备中产生用于因特网协议语音(VoIP)应用的填充帧的方法和装置。 该方法可以包括确定接收到的通信中是否存在丢失帧,其中如果确定存在丢失帧,则设置帧丢失标志并将帧丢失标志存储在帧丢失历史寄存器中,将丢失历史 寄存器,线谱频率(LSF)历史寄存器,语音切换(VCUT)历史寄存器,音调历史寄存器和均方根(RMS)增益历史寄存器,其中丢失历史寄存器,LSF历史寄存器, VCUT历史寄存器,音调历史寄存器和RMS历史寄存器包括至少三个寄存器,三个寄存器是最新的,中间的和最旧的寄存器,将帧丢失标志读入最新的丢失历史寄存器,确定 每个LSF历史寄存器的中间寄存器,VCUT历史寄存器,音调历史寄存器和RMS历史寄存器,并将中间寄存器的内容发送到合成器,以生成 输出语音信号。
    • 10. 发明授权
    • Method and apparatus for managing speech decoders in a communication device
    • 用于在通信设备中管理语音解码器的方法和装置
    • US07970603B2
    • 2011-06-28
    • US11940435
    • 2007-11-15
    • Richard L. Zinser, Jr.Martin W. Egan
    • Richard L. Zinser, Jr.Martin W. Egan
    • G10L19/00
    • G10L19/16G10L19/24
    • A method and apparatus that manages speech decoders in a communication device may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.
    • 管理通信设备中的语音解码器的方法和装置可以包括检测传输速率从较高速率到较低速率的变化,解码和移位第一,第二和第三接收到的第一解码器帧参数组,生成第一解码器 从先前移位的帧参数输出音频帧,产生第一,第二和第三解码器音频填充帧,第二解码器是比第一解码器更高的速率解码器,输出第一和第二第二解码器音频填充帧,组合第一解码器 音频帧和具有重叠三角形窗口的第三第二解码器音频填充帧,并且将组合的第一解码器和第二解码器帧输出到音频缓冲器,以便随后向通信设备的用户发送。 在替代实施例中,另一种方法可以包括检测和处理传输速率从较低速率到较高速率的变化。