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    • 1. 发明授权
    • Method and system for computing a cell normalization factor by sharing arithmetic units in a rake receiver to reduce overall implementation area
    • 通过在耙式接收机中共享运算单元以减少总体实现面积来计算小区归一化因子的方法和系统
    • US08644364B2
    • 2014-02-04
    • US12616239
    • 2009-11-11
    • Thirunathan Sutharsan
    • Thirunathan Sutharsan
    • H04B1/69H04B1/707H04B1/713
    • H04B1/7107H04B1/7115
    • A mobile device receives downlink transmissions comprising replicas of an original downlink transmitted signal over corresponding fingers of a RAKE receiver comprising arithmetic units. The RAKE receiver computes a cell normalization factor for each of active cells and neighbor cells associated with the RAKE receiver. The RAKE receiver uses the same arithmetic units comprising one adder, one multiplier, one divider and/or one square root unit to compute cell normalization factors. The received downlink transmitted signal is processed using the computed cell normalization factors. The RAKE receiver determines signal power from each of other cells, separately, to compute cell normalization factors to normalize fingers of the RAKE receiver. Interference over the normalized fingers are cancelled and used to process the received downlink transmitted signal, which are combined and Turbo decoded. Phase correction is performed over interference cancelled fingers for active cells, but need not be performed for neighbor cells.
    • 移动设备通过包括运算单元的RAKE接收机的相应指状物接收包括原始下行链路发送信号的副本的下行链路传输。 RAKE接收机计算与RAKE接收机相关联的每个活动小区和相邻小区的小区归一化因子。 RAKE接收机使用包括一个加法器,一个乘法器,一个除法器和/或一个平方根单元的相同的算术单元来计算单元格归一化因子。 使用所计算的小区归一化因子来处理所接收的下行链路发送信号。 RAKE接收机分别确定来自每个其他小区的信号功率,以计算细胞归一化因子以使RAKE接收机的手指正常化。 对归一化指状物的干扰被消除并用于处理接收到的下行链路传输信号,它们被组合和Turbo解码。 对于活动单元的干扰消除手指执行相位校正,但不需要对相邻单元执行相位校正。
    • 2. 发明授权
    • Method and system for a programmable interference suppression module
    • 可编程干扰抑制模块的方法和系统
    • US08208856B2
    • 2012-06-26
    • US12686623
    • 2010-01-13
    • Mark HahmWei LuoThirunathan SutharsanAndrew du PreezBin LiuJun WuSeverine Catreux-ErcegShuangquan Wang
    • Mark HahmWei LuoThirunathan SutharsanAndrew du PreezBin LiuJun WuSeverine Catreux-ErcegShuangquan Wang
    • H04B1/00
    • H04B1/71075H04B1/7115H04B1/712H04B2201/70702
    • Aspects of a method and system for a programmable interference suppression module may include receiving a communication signal comprising one or more desired signal, and one or more undesired signals. The communication signal may be utilized to generate estimated channel state information. The estimated channel state information may be formatted for use in interference suppression. A reduced interference signal may be generated from a delayed version of said communications signal and the estimated channel state information, wherein the one or more undesired signals may be attenuated. The reduced interference signal may be formatted for post-processing. The desired signals may comprise WCDMA and/or HSDPA signals, and the undesired signals may be inter-cell and/or intra-cell interference. Further processing may comprise HSDPA processing and/or RAKE finger processing. The communication signal may be a Universal Mobile Telecommunication System (UMTS) compliant signal.
    • 用于可编程干扰抑制模块的方法和系统的方面可以包括接收包括一个或多个期望信号的通信信号以及一个或多个不需要的信号。 通信信号可用于产生估计的信道状态信息。 估计的信道状态信息可以被格式化以用于干扰抑制。 可以从所述通信信号的延迟版本和所估计的信道状态信息产生减小的干扰信号,其中所述一个或多个不需要的信号可以被衰减。 减小的干扰信号可以被格式化用于后处理。 期望的信号可以包括WCDMA和/或HSDPA信号,并且不期望的信号可以是小区间和/或小区内干扰。 进一步的处理可以包括HSDPA处理和/或RAKE手指处理。 通信信号可以是符合通用移动通信系统(UMTS)的信号。
    • 3. 发明申请
    • METHOD AND SYSTEM FOR CHANNEL ESTIMATION PROCESSING FOR INTERFERENCE SUPPRESSION
    • 用于干扰抑制的信道估计处理方法和系统
    • US20110111761A1
    • 2011-05-12
    • US12615237
    • 2009-11-09
    • Wei LuoPan LiuThirunathan Sutharsan
    • Wei LuoPan LiuThirunathan Sutharsan
    • H04B15/00H04W72/04
    • H04L27/00H04B1/7107H04B1/7115H04L25/0204
    • Aspects of a method and system for channel estimation for interference suppression are provided. In this regard, one or more circuits and/or processors of a mobile communication device may generate and/or receive a first set of channel estimates and a second set of channel estimates. The one or more circuits and/or processors may modify the second set of channel estimates based on a comparison of a measure of correlation between the first set of channel estimates and the second set of channel estimates with a threshold. The first set of channel estimates and/or the modified second set of channel estimates may be utilized for cancelling interference in received signals. The first set of channel estimates may be associated with a first transmit antenna of a base transceiver station and the second set of channel estimates may be associated with a second transmit antenna of the base transceiver station.
    • 提供了用于干扰抑制的信道估计的方法和系统的方面。 在这点上,移动通信设备的一个或多个电路和/或处理器可以生成和/或接收第一组信道估计和第二组信道估计。 一个或多个电路和/或处理器可以基于第一组信道估计与具有阈值的第二组信道估计之间的相关度的比较来修改第二组信道估计。 可以利用第一组信道估计和/或修改的第二组信道估计来消除接收信号中的干扰。 第一组信道估计可以与基站收发台的第一发射天线相关联,并且第二组信道估计可以与基站的第二发射天线相关联。
    • 4. 发明申请
    • METHOD AND SYSTEM FOR PROCESSING HIGH QUALITY AUDIO IN A HARDWARE AUDIO CODEC FOR AUDIO TRANSMISSION
    • 用于音频传输的硬件音频编解码器中处理高质量音频的方法和系统
    • US20100057228A1
    • 2010-03-04
    • US12248608
    • 2008-10-09
    • Hongwei KongThirunathan Sutharsan
    • Hongwei KongThirunathan Sutharsan
    • G06F17/00
    • G10L19/02H03H17/0671
    • Aspects of a method and/or system for processing high quality audio in a hardware audio CODEC for audio transmission are provided. In this regard, an audio signal may be down-sampled via a cascaded plurality of filters and sample rate converters in the hardware audio CODEC. Additionally, a portion of each sample of the audio signal may be selected based on an origin of the audio signal. The selected portion of each sample of the audio signal may comprise 16 or 18 bits. The selected portion may be determined based on a type, a class, a manufacturer identifier, and/or a model identifier of the origin the audio signal. Coefficients of the filters may be configured based on the origin of the audio signal. One or more of the filters may comprise one or more cascaded biquads. The sample rate converters may comprise one or more CIC decimation filters.
    • 提供了用于在用于音频传输的硬件音频CODEC中处理高质量音频的方法和/或系统的方面。 在这方面,可以通过硬件音频CODEC中的级联多个滤波器和采样率转换器对音频信号进行下采样。 此外,可以基于音频信号的原点来选择音频信号的每个采样的一部分。 音频信号的每个采样的选定部分可以包括16或18位。 可以基于类型,类别,制造商标识符和/或原始音频信号的模型标识符来确定所选择的部分。 可以基于音频信号的原点配置滤波器的系数。 一个或多个过滤器可以包括一个或多个级联的二重叠。 采样率转换器可以包括一个或多个CIC抽取滤波器。
    • 7. 发明申请
    • METHOD AND SYSTEM FOR DUAL VOICE PATH PROCESSING IN AN AUDIO CODEC
    • 用于音频编解码器中双声道路径处理的方法和系统
    • US20100057473A1
    • 2010-03-04
    • US12248533
    • 2008-10-09
    • Hongwei KongThirunathan Sutharsan
    • Hongwei KongThirunathan Sutharsan
    • G10L19/00
    • G10L19/18
    • Aspects of a method and system for dual voice path processing in an audio CODEC may enable selecting two or more signals received via one or more audio input devices, and filtering and down-sampling each of the selected signals via two or more signal processing branches. Furthermore, an output sample rate of each of the signal processing branches may be configured independently. The signal processing branches may each comprise one or more IIR filters with configurable coefficients and one or more cascaded-integrate-comb (CIC) decimation filters having a configurable decimation ratio. A first of the selected signals, may be filtered and/or down-sampled to generate a signal having a first, lower, sample rate, and a second of the signals may be filtered and/or down-sampled to generate a signal having a second, higher, sample rate. One or more post-processing algorithms such as audio beamforming may also be performed on the selected signals.
    • 用于音频CODEC中的双语音路径处理的方法和系统的方面可以使得能够选择经由一个或多个音频输入设备接收的两个或更多个信号,并且经由两个或更多个信号处理分支对每个所选择的信号进行滤波和下采样。 此外,可以独立地配置每个信号处理分支的输出采样率。 信号处理分支可以各自包括具有可配置系数的一个或多个IIR滤波器和具有可配置抽取比的一个或多个级联积分梳状(CIC)抽取滤波器。 所选择的信号中的第一个可以被滤波和/或下采样以产生具有第一,较低采样率的信号,并且信号中的第二信号可以被滤波和/或下采样以产生具有 第二,较高的采样率。 还可以对所选择的信号执行一个或多个后处理算法,例如音频波束形成。
    • 8. 发明授权
    • Method and system for processing high quality audio in a hardware audio codec for audio transmission
    • 用于处理用于音频传输的硬件音频编解码器中的高质量音频的方法和系统
    • US08909361B2
    • 2014-12-09
    • US12248608
    • 2008-10-09
    • Hongwei KongThirunathan Sutharsan
    • Hongwei KongThirunathan Sutharsan
    • G06F17/00G10L19/02H03H17/06
    • G10L19/02H03H17/0671
    • Aspects of a method and/or system for processing high quality audio in a hardware audio CODEC for audio transmission are provided. In this regard, an audio signal may be down-sampled via a cascaded plurality of filters and sample rate converters in the hardware audio CODEC. Additionally, a portion of each sample of the audio signal may be selected based on an origin of the audio signal. The selected portion of each sample of the audio signal may comprise 16 or 18 bits. The selected portion may be determined based on a type, a class, a manufacturer identifier, and/or a model identifier of the origin the audio signal. Coefficients of the filters may be configured based on the origin of the audio signal. One or more of the filters may comprise one or more cascaded biquads. The sample rate converters may comprise one or more CIC decimation filters.
    • 提供了用于在用于音频传输的硬件音频CODEC中处理高质量音频的方法和/或系统的方面。 在这方面,可以通过硬件音频CODEC中的级联多个滤波器和采样率转换器对音频信号进行下采样。 此外,可以基于音频信号的原点来选择音频信号的每个采样的一部分。 音频信号的每个采样的选定部分可以包括16或18位。 可以基于类型,类别,制造商标识符和/或原始音频信号的模型标识符来确定所选择的部分。 可以基于音频信号的原点配置滤波器的系数。 一个或多个过滤器可以包括一个或多个级联的二重叠。 采样率转换器可以包括一个或多个CIC抽取滤波器。
    • 9. 发明授权
    • Method and system for channel estimation processing for interference suppression
    • 用于干扰抑制的信道估计处理方法及系统
    • US08358610B2
    • 2013-01-22
    • US12615237
    • 2009-11-09
    • Wei LuoPan LiuThirunathan Sutharsan
    • Wei LuoPan LiuThirunathan Sutharsan
    • H04B7/185
    • H04L27/00H04B1/7107H04B1/7115H04L25/0204
    • Aspects of a method and system for channel estimation for interference suppression are provided. In this regard, one or more circuits and/or processors of a mobile communication device may generate and/or receive a first set of channel estimates and a second set of channel estimates. The one or more circuits and/or processors may modify the second set of channel estimates based on a comparison of a measure of correlation between the first set of channel estimates and the second set of channel estimates with a threshold. The first set of channel estimates and/or the modified second set of channel estimates may be utilized for cancelling interference in received signals. The first set of channel estimates may be associated with a first transmit antenna of a base transceiver station and the second set of channel estimates may be associated with a second transmit antenna of the base transceiver station.
    • 提供了用于干扰抑制的信道估计的方法和系统的方面。 在这点上,移动通信设备的一个或多个电路和/或处理器可以生成和/或接收第一组信道估计和第二组信道估计。 一个或多个电路和/或处理器可以基于第一组信道估计与具有阈值的第二组信道估计之间的相关度的比较来修改第二组信道估计。 可以利用第一组信道估计和/或修改的第二组信道估计来消除接收信号中的干扰。 第一组信道估计可以与基站收发台的第一发射天线相关联,并且第二组信道估计可以与基站的第二发射天线相关联。
    • 10. 发明申请
    • Method and System For a Pipelined Dual Audio Path Processing Audio Codec
    • 用于流水线双音频路径处理音频编解码器的方法和系统
    • US20110103593A1
    • 2011-05-05
    • US12613278
    • 2009-11-05
    • Thirunathan Sutharsan
    • Thirunathan Sutharsan
    • H04R5/00G10L21/00G06F17/00
    • G06F3/162H03H17/04H03H17/0664H03H17/0671H03H2218/08H03H2218/085
    • Methods and systems for a pipelined dual audio path processing audio CODEC are disclosed and may comprise centrally generating multiplexer (MUX) select signals for clock domains in an audio CODEC including a plurality of audio inputs and audio processing paths. The MUX select signals may be generated in a single clock domain. Each of the audio processing paths may traverse a plurality of clock domains and may include infinite impulse response (IIR) and cascaded integrator comb (CIC) filters. One or more adders may be shared in the CIC filters, and one or more multipliers and one or more adders may be shared in the IIR filters. The clock domains may be synchronized utilizing the centrally generated enable signals. An output signal of the IIR filters may be buffered in each of the audio paths utilizing a first-in-first-out buffer. The MUX select signals may be generated utilizing a finite state machine.
    • 公开了流水线式双音频路径处理音频CODEC的方法和系统,并且可以包括在包括多个音频输入和音频处理路径的音频编解码器中集中产生用于时钟域的多路复用器(MUX)选择信号。 MUX选择信号可以在单个时钟域中产生。 每个音频处理路径可以遍历多个时钟域,并且可以包括无限脉冲响应(IIR)和级联积分器梳(CIC)滤波器。 一个或多个加法器可以在CIC滤波器中共享,并且一个或多个乘法器和一个或多个加法器可以在IIR滤波器中共享。 可以使用集中产生的使能信号来同步时钟域。 IIR滤波器的输出信号可以利用先进先出缓冲器缓冲在每个音频路径中。 MUX选择信号可以利用有限状态机产生。