LPC-based DTMF receiver for secondary signalling
    1.
    发明授权
    LPC-based DTMF receiver for secondary signalling 失效
    基于LPC的DTMF接收机用于二次信令

    公开(公告)号:US4853958A

    公开(公告)日:1989-08-01

    申请号:US201154

    申请日:1988-06-02

    IPC分类号: H04Q1/457

    CPC分类号: H04Q1/4575

    摘要: A dual tone multi-frequency (DTMF) receiver for use in detecting digitally transmitted signals in the telephone industry. The receiver derives linear predictive coefficients for the digital signals in the data frame. This information is used to compute frequency response magnitudes at the DTMF frequencies. A plurality of magnitude comparisons are then performed to verify the presence of true DTMF signals and concurrently the frequency of these signals is obtained.

    摘要翻译: 一种双音多频(DTMF)接收机,用于检测电话行业中的数字传输信号。 接收器为数据帧中的数字信号导出线性预测系数。 该信息用于计算DTMF频率的频率响应幅度。 然后执行多个幅度比较以验证真实DTMF信号的存在,同时获得这些信号的频率。

    Method and apparatus for improving the voice quality of tandemed vocoders
    2.
    发明授权
    Method and apparatus for improving the voice quality of tandemed vocoders 失效
    提高串联声码器语音质量的方法和装置

    公开(公告)号:US5995923A

    公开(公告)日:1999-11-30

    申请号:US883353

    申请日:1997-06-26

    CPC分类号: G10L19/16 H04W88/181

    摘要: 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.

    摘要翻译: 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。

    Speech bandwidth extension method and apparatus
    3.
    发明授权
    Speech bandwidth extension method and apparatus 失效
    语音带宽扩展方法和装置

    公开(公告)号:US5455888A

    公开(公告)日:1995-10-03

    申请号:US985418

    申请日:1992-12-04

    摘要: 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.

    摘要翻译: 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。

    Joint scheduling of multiple processes on a shared processor
    4.
    发明授权
    Joint scheduling of multiple processes on a shared processor 有权
    在共享处理器上联合调度多个进程

    公开(公告)号:US09098331B2

    公开(公告)日:2015-08-04

    申请号:US13171848

    申请日:2011-06-29

    IPC分类号: G06F9/46 G06F9/48 H04L12/70

    CPC分类号: G06F9/4887 H04L47/00

    摘要: A multi-process scheduler applies a joint optimization criterion to jointly schedule multiple processes executed on a shared processor. The scheduler determines, for each one of a plurality of processes having a predetermined processing time, at least one of an expected arrival time for input data and required delivery time for output data. The scheduler jointly determines process activation times for the processes based on said arrival/delivery, and the processing times, to meet a predetermined joint optimization criterion for the processes. The processes are scheduled on the shared processor according to the jointly determined activation times to minimize queuing delay.

    摘要翻译: 多进程调度器应用联合优化标准来共同调度在共享处理器上执行的多个进程。 对于具有预定处理时间的多个处理中的每一个,调度器确定输入数据的预期到达时间和输出数据的所需传送时间中的至少一个。 调度器基于所述到达/传送和处理时间联合确定用于进程的处理激活时间,以满足用于处理的预定联合优化准则。 这些进程根据联合确定的激活时间在共享处理器上调度以最小化排队延迟。

    MECHANISM FOR DYNAMIC SIGNALING OF ENCODER CAPABILITIES
    5.
    发明申请
    MECHANISM FOR DYNAMIC SIGNALING OF ENCODER CAPABILITIES 有权
    编码器功能动态信号的机制

    公开(公告)号:US20130046534A1

    公开(公告)日:2013-02-21

    申请号:US13588445

    申请日:2012-08-17

    IPC分类号: G10L19/12

    摘要: 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.

    摘要翻译: 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。

    Joint Scheduling of Multiple Processes on a Shared Processor
    6.
    发明申请
    Joint Scheduling of Multiple Processes on a Shared Processor 有权
    共享处理器上多个进程的联合调度

    公开(公告)号:US20130007754A1

    公开(公告)日:2013-01-03

    申请号:US13171848

    申请日:2011-06-29

    IPC分类号: G06F9/46

    CPC分类号: G06F9/4887 H04L47/00

    摘要: A multi-process scheduler applies a joint optimization criterion to jointly schedule multiple processes executed on a shared processor. The scheduler determines, for each one of a plurality of processes having a predetermined processing time, at least one of an expected arrival time for input data and required delivery time for output data. The scheduler jointly determines process activation times for the processes based on said arrival/delivery, and the processing times, to meet a predetermined joint optimization criterion for the processes. The processes are scheduled on the shared processor according to the jointly determined activation times to minimize queuing delay.

    摘要翻译: 多进程调度器应用联合优化标准来共同调度在共享处理器上执行的多个进程。 对于具有预定处理时间的多个处理中的每一个,调度器确定输入数据的预期到达时间和输出数据的所需传送时间中的至少一个。 调度器基于所述到达/传送和处理时间联合确定用于进程的处理激活时间,以满足用于处理的预定联合优化准则。 这些进程根据联合确定的激活时间在共享处理器上调度以最小化排队延迟。

    METHOD AND APPARATUS FOR TIME ALIGNMENT ALONG A MULTI-NODE COMMUNICATION LINK
    7.
    发明申请
    METHOD AND APPARATUS FOR TIME ALIGNMENT ALONG A MULTI-NODE COMMUNICATION LINK 有权
    多节点通信链路上时间对齐的方法与装置

    公开(公告)号:US20090046698A1

    公开(公告)日:2009-02-19

    申请号:US11839861

    申请日:2007-08-16

    IPC分类号: H04J3/06

    摘要: 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.

    摘要翻译: 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。

    System for TDMA mobile-to-mobile VSELP CODEC bypass
    8.
    发明授权
    System for TDMA mobile-to-mobile VSELP CODEC bypass 失效
    用于TDMA移动到移动VSELP CODEC旁路的系统

    公开(公告)号:US06185424B2

    公开(公告)日:2001-02-06

    申请号:US09096192

    申请日:1998-06-12

    IPC分类号: H04Q722

    摘要: In a TDMA mobile-to-mobile connection, the end-to-end audio signal quality as well as system performance can be improved by providing digital signal processors the capability to automatically switch configuration such that each digital signal processor in a mobile-to-mobile communication connection can automatically identify a TDMA mobile-to-mobile connection and bypass the speech encoding and decoding processes within the digital signal processors. The two digital signal processors are virtually connected at the channel codecs.

    摘要翻译: 在TDMA移动到移动连接中,可以通过向数字信号处理器提供自动切换配置的能力来提高端对端音频信号质量以及系统性能,使得每个数字信号处理器以移动 - 移动通信连接可以自动识别TDMA移动到移动连接并且绕过数字信号处理器内的语音编码和解码过程。 两个数字信号处理器在通道编解码器上虚拟连接。

    Methods and apparatus for echo suppression
    9.
    发明授权
    Methods and apparatus for echo suppression 失效
    用于回波抑制的方法和装置

    公开(公告)号:US6011846A

    公开(公告)日:2000-01-04

    申请号:US881062

    申请日:1997-06-24

    摘要: 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.

    摘要翻译: 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。