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    • 1. 发明申请
    • FRAME ERASURE CONCEALMENT FOR A MULTI RATE SPEECH AND AUDIO CODEC
    • 多语音和音频编解码器的帧擦除保护
    • US20120265523A1
    • 2012-10-18
    • US13443204
    • 2012-04-10
    • Steven Craig GREERHosang Sung
    • Steven Craig GREERHosang Sung
    • G10L21/00G10L19/00
    • G10L19/005G10L19/002G10L19/24
    • An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec, configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode, the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.
    • 提供一种音频编码终端和方法。 该终端包括编码模式设置单元,用于根据多个操作模式设置用于通过编解码器进行输入音频编码的操作模式,用于基于所设置的操作模式对输入音频进行编码,使得当所设置的操作模式为高帧时 擦除率(FER)模式,编解码器根据一种或多种FEC模式的选择帧擦除隐藏(FEC)模式对输入音频的当前帧进行编码。 在将操作模式设置为高FER模式时,从针对高FER模式预定的一个或多个FEC模式中选择一个FEC模式,以通过在输入音频的编码中并入冗余来控制编解码器 或者根据所选择的一种FEC模式作为单独的冗余信息与编码的输入音频分离。
    • 3. 发明授权
    • Self-calibrating frequency standard system
    • 自校准频率标准系统
    • US5697082A
    • 1997-12-09
    • US525111
    • 1995-09-08
    • Steven Craig GreerHussein S. El-Ghoroury
    • Steven Craig GreerHussein S. El-Ghoroury
    • G04G7/00H03L7/00H04L7/00H04B7/19
    • H03L7/00G04G7/00
    • A system for self-calibrating a clock of a communication terminal for use with communication systems in which a central communication node generates time base correction signals for the terminal clock includes a terminal oscillator which generates an oscillator frequency that includes an error amount. An oscillator calibration filter generates a frequency error estimate amount. Circuitry is provided for subtracting the frequency error estimate amount generated by the calibration filter from the oscillator frequency error amount. Circuitry is provided for applying the time base correction signals to the calibration filter to thereby modify the frequency error estimate amount generated by the calibration filter based upon the time base correction signals generated by the communication central node.
    • 一种用于自校准用于与中央通信节点产生用于终端时钟的时基校正信号的通信系统一起使用的通信终端的时钟的系统包括产生包括错误量的振荡器频率的终端振荡器。 振荡器校准滤波器产生频率误差估计量。 电路用于从振荡器频率误差量减去由校准滤波器产生的频率误差估计量。 提供电路用于将时基校正信号施加到校准滤波器,从而基于由通信中心节点产生的时基校正信号来修改由校准滤波器产生的频率误差估计量。
    • 4. 发明授权
    • Frame erasure concealment for a multi rate speech and audio codec
    • 用于多速率语音和音频编解码器的帧擦除隐藏
    • US09026434B2
    • 2015-05-05
    • US13443204
    • 2012-04-10
    • Steven Craig GreerHosang Sung
    • Steven Craig GreerHosang Sung
    • G10L19/00G10L19/005G10L19/24
    • G10L19/005G10L19/002G10L19/24
    • An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.
    • 提供一种音频编码终端和方法。 终端包括编码模式设置单元,用于根据多个操作模式设置用于通过编解码器进行输入音频编码的操作模式,编解码器被配置为基于设置的操作模式对输入音频进行编码,使得当设置的操作模式是高帧擦除 速率(FER)模式,编解码器根据一个或多个FEC模式的选择帧擦除隐藏(FEC)模式对输入音频的当前帧进行编码。 在将操作模式设置为高FER模式时,从针对高FER模式预定的一个或多个FEC模式中选择一个FEC模式,以通过在输入音频的编码中并入冗余来控制编解码器,或 作为根据所选择的一种FEC模式与编码的输入音频分离的单独的冗余信息。
    • 5. 发明授权
    • Codec-assisted capacity enhancement of wireless VoIP
    • 无线VoIP的编解码器辅助容量增强
    • US07546508B2
    • 2009-06-09
    • US10740863
    • 2003-12-19
    • Steven Craig Greer
    • Steven Craig Greer
    • G08C25/02H04L1/18
    • G10L19/005G10L19/16H04W88/181
    • The use of a cellular system's packet data channel for transport of speech communications is enhanced by dynamically controlling the transmission of data packets that were received in error and notified to the transmitter by the receiver in accordance with an algorithm that takes into account a self-generated determination of the perceptual quality importance of the frame and is generated once for each frame by the speech encoder. The calculation of the frame importance is codec-dependent, and depends on a number of attributes associated with each particular frame, including the frame rate (for variable-rate codecs), signal classification, transition characteristics, background noise, previous frame erasures, etc. The selective reduction in transmission and retransmission increases the capacity of the cellular system while maintaining or even improving the quality of the received voice signal.
    • 使用蜂窝系统的分组数据信道进行语音通信的传输通过动态地控制错误接收的数据分组的传输并且由接收机根据考虑到自生成的算法而被发送给发射机的传输被增强 确定帧的感知质量重要性,并且由语音编码器针对每个帧产生一次。 帧重要性的计算取决于编解码,并且取决于与每个特定帧相关联的多个属性,包括帧速率(用于可变速率编解码器),信号分类,转换特性,背景噪声,先前帧擦除等 传输和重传的选择性降低增加了蜂窝系统的容量,同时维持或甚至提高了接收到的语音信号的质量。
    • 6. 发明授权
    • Apparatus and method for efficiently supporting VoIP in a wireless communication system
    • 在无线通信系统中有效支持VoIP的装置和方法
    • US07911945B2
    • 2011-03-22
    • US10918542
    • 2004-08-12
    • Zhigang RongLin MaSteven Craig GreerZhigang LiuZhouyue Pi
    • Zhigang RongLin MaSteven Craig GreerZhigang LiuZhouyue Pi
    • G08C15/00
    • H04L65/1069H04L29/06027H04L65/80H04W28/06H04W80/00
    • The present invention concerns methods for efficiently supporting Voice over Internet Protocol (VoIP) on the Forward Packet Data Channel (F-PDCH) in CDMA 2000 1xEV-DV systems. Active speech in VoIP is encoded using, for example enhanced variable rate codec (EVRC), which produces 171, 80 and 16 bits per 20 ms of speech for Rate 1, Rate ½ and Rate ⅛, respectively. However, about 60% of the time a user is inactive during a speech session, so an inordinate amount of system bandwidth is comprised of rate ⅛ VoIP packets. In one embodiment of the present invention the apparatus of the present invention identifies the Rate ⅛ voice frame packets and discards them. In another embodiment of the present invention, the apparatus of the present invention identifies the Rate ⅛ voice frame packets and selects some of them for further transmission. In both embodiments the efficiency of channel utilization is increased since the amount of channel band width used to communicate relatively little information, e.g., gaps of silence, is decreased.
    • 本发明涉及在CDMA 2000 1xEV-DV系统中有效支持前向分组数据信道(F-PDCH)上的因特网协议语音(VoIP)的方法。 VoIP中的活动语音使用例如增强型可变速率编解码器(EVRC)进行编码,其分别为速率1,速率½和速率per分别产生每20毫秒语音的171,80和16比特。 然而,在用户在语音会话期间不活动的时间的大约60%,因此系统带宽的过量由速率⅛VoIP分组组成。 在本发明的一个实施例中,本发明的装置识别速率⅛语音帧分组并丢弃它们。 在本发明的另一实施例中,本发明的装置识别速率⅛语音帧分组,并选择其中一些用于进一步传输。 在两个实施例中,信道利用的效率增加,因为用于传达相对较少的信息的信道带宽的量,例如,沉默的间隙被减少。