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    • 1. 发明授权
    • Speech bandwidth extension method and apparatus
    • 语音带宽扩展方法和装置
    • US5455888A
    • 1995-10-03
    • US985418
    • 1992-12-04
    • Vasu IyengarRafi RabipourPaul MermelsteinBrian R. Shelton
    • Vasu IyengarRafi RabipourPaul MermelsteinBrian R. Shelton
    • G10L19/00G10L19/02G10L21/02G10L5/06
    • G10L21/038G10L19/0204G10L21/0232
    • A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.
    • 语音带宽扩展方法和装置分析使用LPC分析以8kHz采样的窄带语音,以确定其频谱形状和反向滤波以提取其激励信号。 激励信号内插到16 kHz的采样率,并分析了音调控制和功率电平。 然后对产生白噪声的宽带信号进行滤波以提供合成的宽带激励信号。 确定窄带形状并将其与各矢量量化器码本中的模板进行比较,以选择各自的高频带形状和增益。 然后对合成的宽带激励信号进行滤波以提供高频带信号,该高频带信号又被加到窄带信号中,以16kHz采样率内插,以产生人造宽带信号。 该装置可以在数字信号处理器芯片上实现。
    • 2. 发明授权
    • Methods and apparatus for estimating and adjusting the frequency
response of telecommunications channels
    • 用于估计和调整电信频道频率响应的方法和装置
    • US5577117A
    • 1996-11-19
    • US257129
    • 1994-06-09
    • Rafi RabipourVasu Iyengar
    • Rafi RabipourVasu Iyengar
    • H04B3/06H04R29/00
    • H04B3/06H04L25/03885
    • In methods and apparatus for estimating the frequency response of telecommunications channels, signals carried on the channels are divided into pluralities of signal segments of limited duration, unvoiced signal segments are identified, and spectral components of the unvoiced signal segments are measured. The measured spectral components may be integrated over time to derive time-averaged measurements of the spectral components. The time-averaged measurements may be compared to corresponding components of expected unvoiced signal spectra to improve the accuracy of the frequency response estimates. Filters may be inserted in the channels to adjust the frequency responses, the filters having filter characteristics selected in response to the measured spectral components. The methods and apparatus can be used to compensate for poor bass frequency response in voice channels.
    • 在用于估计电信信道的频率响应的方法和装置中,在信道上承载的信号被划分成多个有限持续时间的信号段,识别清音信号段,并且测量清音信号段的频谱分量。 测量的光谱分量可以随时间积分以导出光谱分量的时间平均测量。 可以将时间平均测量值与预期清音信号频谱的相应分量进行比较,以提高频率响应估计的准确度。 可以将滤波器插入通道中以调整频率响应,滤波器具有响应于测量的频谱分量而选择的滤波器特性。 方法和装置可用于补偿语音信道中的低音低频响应。
    • 3. 发明授权
    • Method and apparatus for improving the voice quality of tandemed vocoders
    • 提高串联声码器语音质量的方法和装置
    • US5995923A
    • 1999-11-30
    • US883353
    • 1997-06-26
    • Paul MermelsteinRafi RabipourWilliam NavarroPaul Coverdale
    • Paul MermelsteinRafi RabipourWilliam NavarroPaul Coverdale
    • G10L19/02G10L19/14H03M7/30H04W88/18H03M9/00
    • G10L19/16H04W88/181
    • In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.
    • 近年来,电讯业目睹了各种数字声码器的扩散,以满足不同有线和无线通讯系统的带宽需求。 网络的多样性的快速增长和这种网络的用户数量正在增加两个声码器串联放置以服务于单个连接的情况的数量。 低比特率编解码器的这种布置可能会降低发送语音的质量。 为了克服这个问题,本发明提供了一种用于在无线通信环境中发送数字化语音信号的新颖方法和装置。 该装置能够通过中间公共格式将压缩的语音信号从一种格式转换成另一格式,从而避免必须将语音数据连续解压缩到PCM型数字化,然后再重新压缩语音数据。
    • 6. 发明申请
    • MECHANISM FOR DYNAMIC SIGNALING OF ENCODER CAPABILITIES
    • 编码器功能动态信号的机制
    • US20130046534A1
    • 2013-02-21
    • US13588445
    • 2012-08-17
    • Rafi RabipourChung-Cheung ChuDaniel Cohn
    • Rafi RabipourChung-Cheung ChuDaniel Cohn
    • G10L19/12
    • H04M7/0072G10L19/00H04L65/1059H04L65/608H04W28/18H04W88/181
    • The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node.
    • 本公开提供用于动态地信令相应通信节点的声码器的编码器能力的系统和方法。 在一个实施例中,在第一通信节点和第二通信节点之间的呼叫期间,用于第一通信节点的控制节点(例如,基站控制器或移动交换中心)发送用于第一通信节点的声码器的语音编码器的能力信息 通信节点到第二通信节点的控制节点。 结果,第二通信节点能够基于第一通信节点的声码器的语音编码器的能力来选择并请求第一通信节点的声码器的语音编码器的优选编码器模式。
    • 8. 发明申请
    • METHOD AND APPARATUS FOR TIME ALIGNMENT ALONG A MULTI-NODE COMMUNICATION LINK
    • 多节点通信链路上时间对齐的方法与装置
    • US20090046698A1
    • 2009-02-19
    • US11839861
    • 2007-08-16
    • Chung Cheung CHURafi Rabipour
    • Chung Cheung CHURafi Rabipour
    • H04J3/06
    • H04L12/66H04N21/2335H04N21/23805H04N21/6181H04N21/64307H04N21/64322
    • A network entity, which comprises an input configured to receive from an upstream network entity a stream of first media data elements; an output configured to release towards a downstream network entity a stream of second media data elements; a processing engine configured to effect processing tasks on the first media data elements, thereby to generate the second media data elements, the processing tasks being effected in a set of processing intervals; and a control entity. The control entity is configured for receiving a request for a first phase adjustment from the downstream network entity; modifying the set of processing intervals in which are effected the processing tasks in an attempt to accommodate the first phase adjustment; determining a second phase adjustment based on arrival characteristics of the first media data elements and the modified set of processing intervals; and releasing towards the upstream network entity a request for the second phase adjustment.
    • 网络实体,其包括被配置为从上游网络实体接收第一媒体数据元素流的输入; 被配置为向下游网络实体释放第二媒体数据元素流的输出; 处理引擎被配置为对所述第一媒体数据元素执行处理任务,从而生成所述第二媒体数据元素,所述处理任务在一组处理间隔中进行; 和控制实体。 控制实体被配置为从下游网络实体接收对第一阶段调整的请求; 修改处理间隔的集合,以试图适应第一阶段调整; 基于所述第一媒体数据元素的到达特性和所述经修改的处理间隔集确定第二相位调整; 向上游网络实体发布第二阶段调整请求。
    • 10. 发明授权
    • Methods and apparatus for echo suppression
    • 用于回波抑制的方法和装置
    • US6011846A
    • 2000-01-04
    • US881062
    • 1997-06-24
    • Rafi RabipourDominic HoMajid FoodeeiMadeleine Saikaly
    • Rafi RabipourDominic HoMajid FoodeeiMadeleine Saikaly
    • G10L21/02H04B3/20H04B7/015G10L9/00
    • H04B7/015H04B3/20G10L2021/02082
    • In methods and apparatus for suppressing echo of a far end signal encoded using LPC-based compression in a near end signal encoded using LPC-based compression, parameters of each frame of the near end encoded signal are processed without synthesizing a speech signal from the near end encoded signal to determine whether sufficient echo to merit echo suppression is present in the frame. Upon determining that insufficient echo to merit echo suppression is present in the frame, the parameters of the frame are passed unmodified. Upon determining that sufficient echo to merit echo suppression is present in said frame, the parameters of the frame are modified without synthesizing a speech signal to suppress echo in the frame. The methods and apparatus are particularly suitable in codec bypass applications.
    • 在使用基于LPC的压缩编码的近端信号中抑制使用基于LPC的压缩编码的远端信号的回波的方法和装置中,处理近端编码信号的每帧的参数,而不从近端 结束编码信号以确定帧中是否存在足以有效回波抑制的回波。 在确定帧中存在不足以获得回波抑制的回波的情况下,帧的参数未被修改。 在确定在所述帧中存在足够的回波以获得回波抑制的情况下,修改帧的参数而不合成语音信号以抑制帧中的回波。 该方法和装置特别适用于编解码器旁路应用。