Method and apparatus for frame classification and rate determination in voice transcoders for telecommunications
    1.
    发明申请
    Method and apparatus for frame classification and rate determination in voice transcoders for telecommunications 失效
    用于电信的语音转码器中的帧分类和速率确定的方法和装置

    公开(公告)号:US20050049855A1

    公开(公告)日:2005-03-03

    申请号:US10642422

    申请日:2003-08-14

    IPC分类号: G10L19/10 G10L19/14

    CPC分类号: G10L19/173 G10L19/10

    摘要: A method and apparatus for frame classification and rate determination in voice transcoders. The apparatus includes a classifier input parameter preparation module that unpacks the bitstream from the source codec and selects the codec parameters to be used for classification, parameter buffers that store previous input and output parameters of previous frames, and a frame classification and rate decision module that uses the source codec parameters from the current frame and zero or more frames to determine the frame class, rate, and classification feature parameters for the destination codec. The classifier input parameter preparation module separates the bitstream code and unquantizes the sub-codes into the codec parameters. These codec parameters may include line spectral frequencies, pitch lag, pitch gains, fixed codebook gains, fixed codebook vectors, rate and frame energy. The frame classification and rate decision module comprises M sub-classifiers and a final decision module. The characteristics of the sub-classifiers are obtained by a classifier construction module, which comprises a training set generation module, a learning module and an evaluation module. The method includes preparing the classifier input parameters, constructing the frame and rate classifier and determining the frame class, rate decision and classification feature parameters for the destination codec using the intermediate parameters and bit rate of the source codec. Constructing the frame and rate classifier includes generating the training and test data and training and/or building the classifier.

    摘要翻译: 一种用于语音转码器中帧分类和速率确定的方法和装置。 该装置包括:分类器输入参数准备模块,用于从源编解码器解码比特流并选择要用于分类的编解码器参数,存储先前帧的先前输入和输出参数的参数缓冲器;以及帧分类和速率决定模块, 使用来自当前帧的源编解码器参数和零个或多个帧来确定目标编解码器的帧类,速率和分类特征参数。 分类器输入参数准备模块分离比特流代码,并将子代码未量化为编解码器参数。 这些编解码器参数可以包括线谱频率,音调滞后,音调增益,固定码本增益,固定码本向量,速率和帧能量。 帧分类和速率决定模块包括M个子分类器和一个最终决策模块。 分类器的特征由分类器构造模块获得,分类器构造模块包括训练集生成模块,学习模块和评估模块。 该方法包括准备分类器输入参数,构建帧和速率分类器,并使用源编解码器的中间参数和比特率确定目标编解码器的帧类,速率决策和分类特征参数。 构建框架和速率分类器包括生成训练和测试数据,训练和/或构建分类器。

    Method and apparatus for voice transcoding between variable rate coders
    2.
    发明申请
    Method and apparatus for voice transcoding between variable rate coders 失效
    用于可变速率编码器之间语音转码的方法和装置

    公开(公告)号:US20050053130A1

    公开(公告)日:2005-03-10

    申请号:US10660468

    申请日:2003-09-10

    IPC分类号: G10L19/12 G10L19/14 H04B1/66

    摘要: A variable rate compressed voice signal domain transcoder that transcodes a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard; the second voice compression standard defines a variable-rate voice codec. The method includes unquantizing a bitstream into a first set of parameters compatible with a first compression standard. The first set of parameters in addition to external control commands are then used to determine the frame class and rate for the second compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard according to the frame-classification and rate determination decision without converting the first set of parameters to an analog or digital voice waveform representation. The transformation approaches can be varied and further optimized based on the characteristics of the pair of first compression standard and the second compression standard. Lastly, the second set of parameters is packed into a bitstream compatible with the second compression standard.

    摘要翻译: 一种可变速率压缩语音信号域代码转换器,其将表示根据第一语音压缩标准编码的数据的比特流代码成表示根据第二语音压缩标准的数据帧的比特流; 第二语音压缩标准定义了可变速率语音编解码器。 所述方法包括将比特流非量化到与第一压缩标准兼容的第一组参数中。 然后,除了外部控制命令之外,第一组参数用于确定第二个压缩标准的帧类和速率。 接下来,根据帧分类和速率确定决定,将第一组参数变换为与第二压缩标准兼容的第二参数集合,而不将第一组参数转换为模拟或数字语音波形表示。 基于第一压缩标准对和第二压缩标准的特性,可以改变变形方式并进一步优化。 最后,将第二组参数打包成与第二压缩标准兼容的比特流。

    METHOD AND APPARATUS FOR VIDEO SERVICES
    3.
    发明申请
    METHOD AND APPARATUS FOR VIDEO SERVICES 审中-公开
    视频服务的方法和设备

    公开(公告)号:US20090232129A1

    公开(公告)日:2009-09-17

    申请号:US12400721

    申请日:2009-03-09

    IPC分类号: H04L12/66

    摘要: A method for providing a multimedia service to a multimedia terminal includes establishing an audio link between the multimedia terminal and a server over an audio channel, and detecting one or more media capabilities of the multimedia terminal. The method also includes providing an application logic for the multimedia service, establishing a visual link between the multimedia terminal and the server over a video channel, providing an audio stream for the multimedia service over the audio link, and providing a visual stream for the multimedia service over the video link. The method further includes combining the video link and the audio link, and adjusting a transmission time of one or more packets in the visual stream to synchronize the visual stream with the audio stream.

    摘要翻译: 用于向多媒体终端提供多媒体服务的方法包括:通过音频信道建立多媒体终端与服务器之间的音频链路,以及检测多媒体终端的一个或多个媒体能力。 该方法还包括提供用于多媒体服务的应用逻辑,通过视频信道建立多媒体终端与服务器之间的可视链接,通过音频链路提供多媒体服务的音频流,以及为多媒体提供视频流 视频链接服务。 该方法还包括组合视频链路和音频链路,以及调整可视流中的一个或多个分组的传输时间以使视频流与音频流同步。

    METHOD AND APPARATUS FOR COMPRESSED VIDEO BITSTREAM CONVERSION WITH REDUCED-ALGORITHMIC-DELAY
    4.
    发明申请
    METHOD AND APPARATUS FOR COMPRESSED VIDEO BITSTREAM CONVERSION WITH REDUCED-ALGORITHMIC-DELAY 有权
    用缩减算法延迟的压缩视频转换的方法和装置

    公开(公告)号:US20080080619A1

    公开(公告)日:2008-04-03

    申请号:US11862117

    申请日:2007-09-26

    IPC分类号: H04B1/66

    摘要: The present invention relates to converting media bitstreams across different networks in a media gateway without any algorithmic delay, and reduces the computation load within the transmission in the situation where the bandwidth of the outgoing network varies dynamically. A first embodiment of the present invention provides an apparatus and a method for a Reduced-Algorithmic-Delay Media Stream Unit Conversion module which is a light weight Simple Pass-Through operation. A second embodiment of the present invention provides an apparatus and a method for a Reduced-Algorithmic-Delay Video Rate Conversion. An alternative embodiment provides an apparatus and a method for a Smart Pass-Through Operation which involves switching between the Simple Pass-Through and the Rate Converter. The methods and apparatuses provided by the first and second embodiment can be used as a stand alone system, or as part of the module of the alternative embodiment.

    摘要翻译: 本发明涉及在媒体网关中的不同网络上转换媒体比特流而没有任何算法延迟,并且在出局网络的带宽动态变化的情况下减少传输内的计算负荷。 本发明的第一实施例提供了一种重量轻的简单通过操作的减速算法延迟媒体流单元转换模块的装置和方法。 本发明的第二实施例提供了一种用于减少算法延迟视频速率转换的装置和方法。 替代实施例提供了一种智能直通操作的装置和方法,其涉及在简单通过和速率转换器之间切换。 由第一和第二实施例提供的方法和装置可以用作独立系统,或者作为替代实施例的模块的一部分。

    Method and System for Video Encoding and Transcoding
    5.
    发明申请
    Method and System for Video Encoding and Transcoding 有权
    视频编码和代码转换的方法和系统

    公开(公告)号:US20070286286A1

    公开(公告)日:2007-12-13

    申请号:US11738816

    申请日:2007-04-23

    IPC分类号: H04B1/66

    摘要: A method of removing a motion vector from a group of motion vectors used in an encoding process includes providing a list of motion vectors, selecting an initial motion vector from the list of motion vectors, and providing an intermediate motion vector using a motion vector refinement process. The motion vector refinement process uses, in part, the initial motion vector. The method also includes forming a region defined by one or more parameters associated with the initial motion vector and one or more parameters associated with the intermediate motion vector, selecting an additional motion vector from the list of motion vectors, determining that the additional motion vector points into the region, and modifying a state of the additional motion vector.

    摘要翻译: 从编码处理中使用的运动矢量组中去除运动矢量的方法包括提供运动矢量列表,从运动矢量列表中选择初始运动矢量,以及使用运动矢量细化处理提供中间运动矢量 。 运动矢量细化过程部分地使用初始运动矢量。 该方法还包括形成由与初始运动矢量相关联的一个或多个参数定义的区域以及与中间运动矢量相关联的一个或多个参数,从运动矢量列表中选择附加运动矢量,确定附加运动矢量点 并且修改附加运动矢量的状态。

    Method and Apparatus for Audio Transcoding
    6.
    发明申请
    Method and Apparatus for Audio Transcoding 有权
    用于音频转码的方法和装置

    公开(公告)号:US20070288234A1

    公开(公告)日:2007-12-13

    申请号:US11738822

    申请日:2007-04-23

    IPC分类号: G10L19/00

    CPC分类号: G10L19/173

    摘要: An apparatus for transcoding an audio signal between a CELP-based coder and a hybrid coder includes a source bitstream unwrapper configured to receive a source bitstream, extract one or more CELP compression parameters from the source bitstream, and construct an audio signal vector from the source bitstream while maintaining the one or more extracted CELP compression parameters. The apparatus also includes a frame interpolator coupled to the source bitstream unwrapper and a compression parameter converter coupled to frame interpolator. The compression parameter converter is configured to calculate output compression parameters from at least one of the interpolated compression parameters or the one or more extracted CELP compression parameters. Additionally, the apparatus includes a destination bitstream wrapper coupled to the compression parameter converter and a mapping parameter tuner coupled to the frame interpolator. The mapping parameter tuner is configured to select one or more parameters for use by the compression parameter converter.

    摘要翻译: 一种用于在基于CELP的编码器和混合编码器之间对音频信号进行代码转换的装置包括:源比特流解包器,被配置为接收源比特流,从源比特流提取一个或多个CELP压缩参数,并从源构建音频信号向量 同时保持一个或多个提取的CELP压缩参数。 该装置还包括耦合到源比特流不包装器的帧内插器和耦合到帧内插器的压缩参数转换器。 压缩参数转换器被配置为从内插的压缩参数或一个或多个提取的CELP压缩参数中的至少一个来计算输出压缩参数。 另外,该装置包括耦合到压缩参数转换器的目的比特流封装器和耦合到帧内插器的映射参数调谐器。 映射参数调谐器被配置为选择一个或多个参数供压缩参数转换器使用。

    Method and apparatus for DTMF detection and voice mixing in the CELP parameter domain
    7.
    发明申请
    Method and apparatus for DTMF detection and voice mixing in the CELP parameter domain 审中-公开
    在CELP参数域中进行DTMF检测和语音混合的方法和装置

    公开(公告)号:US20070025546A1

    公开(公告)日:2007-02-01

    申请号:US11524345

    申请日:2006-09-19

    IPC分类号: H04M3/00

    CPC分类号: G10L19/12 H04L27/30 H04Q1/46

    摘要: A method and apparatus for DTMF detection and voice mixing in the code-excited linear prediction (CELP) parameter space, without fully decoding and reconstructing the speech signal. The apparatus includes a Dual Tone Multiplexed Frequency (DTMF) signal detection module and a multi-input mixing module. The DTMF signal detection module detects DTMF signals by computing characteristic features from the input CELP parameters and comparing them with known features of DTMF signals. The multi-input mixing module mixes multiple sets of input CELP parameters, that represent multiple voice signals, into a single set of CELP parameters. The mixing computation is performed by analyzing each set of input CELP parameters, determining the order of importance of the input sets, selecting a strategy for mixing the CELP parameters, and outputting the mixed CELP parameters. The method includes inputting one or more sets of CELP parameters and external commands, detecting DTMF tones, mixing multiple sets of CELP parameters and outputting the DTMF signal, if detected, and the mixed CELP parameters.

    摘要翻译: 一种在码激励线性预测(CELP)参数空间中用于DTMF检测和语音混合的方法和装置,没有完全解码和重建语音信号。 该装置包括双音多频(DTMF)信号检测模块和多输入混频模块。 DTMF信号检测模块通过从输入CELP参数中计算特征,并与DTMF信号的已知特征进行比较来检测DTMF信号。 多输入混合模块将表示多个语音信号的多组输入CELP参数混合成一组CELP参数。 通过分析每组输入CELP参数,确定输入集的重要性的顺序,选择用于混合CELP参数的策略,并输出混合的CELP参数来执行混合计算。 该方法包括输入一组或多组CELP参数和外部命令,检测DTMF音调,混合多组CELP参数并输出DTMF信号(如果检测到)和混合CELP参数。

    Method and apparatus for voice trans-rating in multi-rate voice coders for telecommunications
    8.
    发明申请
    Method and apparatus for voice trans-rating in multi-rate voice coders for telecommunications 审中-公开
    用于电信的多速率语音编码器中语音传输的方法和装置

    公开(公告)号:US20050258983A1

    公开(公告)日:2005-11-24

    申请号:US10843844

    申请日:2004-05-11

    IPC分类号: G10L19/14 H03M7/00

    CPC分类号: G10L19/173

    摘要: Method and apparatus for trans-rating a bitstream of data through multi-rate voice coders converting a bitstream representing frames of data encoded according to a first voice compression method of a first rate to a second voice compression method according to a second rate. A trans-rating pair includes voice compression parameters mapping modules. The method of trans-rating includes either bit-unpacking or unquantization on an encoded packet at input site to obtain rate information and voice compression parameters according to the first rate voice compression method. The information of the first rate and the required output rate, namely a second rate type, in addition to external control commands, are then used to determine the converting strategy of the trans-rating pair. Next, at least some of the compression parameters of the first rate are passed through, or mapped, into compression parameters of the second rate compatible with the second rate voice compression method.

    摘要翻译: 用于通过多速率语音编码器来转换数据比特流的方法和装置,其将表示根据第一速率的第一语音压缩方法编码的数据的比特流转换为第二语音压缩方法。 变压器对包括语音压缩参数映射模块。 转移方法包括在输入站点的编码数据包上进行位解包或非​​量化,以根据第一速率语音压缩方法获得速率信息和语音压缩参数。 然后使用第一速率和所需输出速率的信息,即外部控制命令之外的第二速率类型,以确定变压器对的转换策略。 接下来,将第一速率的至少一些压缩参数通过或映射成与第二速率语音压缩方法兼容的第二速率的压缩参数。

    Transcoding method and system between CELP-based speech codes with externally provided status
    9.
    发明申请
    Transcoding method and system between CELP-based speech codes with externally provided status 有权
    基于CELP的语音代码与外部提供状态之间的转码方法和系统

    公开(公告)号:US20080077401A1

    公开(公告)日:2008-03-27

    申请号:US11711467

    申请日:2007-02-26

    IPC分类号: G10L19/00

    CPC分类号: G10L19/12 G10L19/173

    摘要: A method for transcoding a CELP based compressed voice bitstream from source codec to destination codec. The method includes processing a source codec input CELP bitstream to unpack at least one or more CELP parameters from the input CELP bitstream and interpolating one or more of the plurality of unpacked CELP parameters from a source codec format to a destination codec format if a difference of one or more of a plurality of destination codec parameters including a frame size, a subframe size, and/or sampling rate of the destination codec format and one or more of a plurality of source codec parameters including a frame size, a subframe size, or sampling rate of the source codec format exist. The method includes encoding the one or more CELP parameters for the destination codec and processing a destination CELP bitstream by at least packing the one or more CELP parameters for the destination codec.

    摘要翻译: 一种用于将基于CELP的压缩语音比特流从源编解码器转码为目标编解码器的方法。 该方法包括:处理源编解码器输入CELP比特流以从输入CELP比特流解压缩至少一个或多个CELP参数,并且将多个未打包的CELP参数中的一个或多个从源编解码器格式内插到目的地编解码器格式,如果 多个目的地编解码器参数中的一个或多个包括目标编解码器格式的帧大小,子帧大小和/或采样率以及多个源编解码器参数中的一个或多个,包括帧大小,子帧大小或 存在源编解码格式的采样率。 该方法包括通过至少打包目的编解码器的一个或多个CELP参数来编码目的编解码器的一个或多个CELP参数并处理目的地CELP比特流。

    METHOD AND APPARATUS OF VOICE MIXING FOR CONFERENCING AMONGST DIVERSE NETWORKS
    10.
    发明申请
    METHOD AND APPARATUS OF VOICE MIXING FOR CONFERENCING AMONGST DIVERSE NETWORKS 失效
    用于配置多个多元网络的语音混合的方法和装置

    公开(公告)号:US20070299661A1

    公开(公告)日:2007-12-27

    申请号:US11564794

    申请日:2006-11-29

    IPC分类号: G10L19/12 G10L19/14 H04L12/18

    CPC分类号: H04M3/568

    摘要: A conferencing system is provided that utilizes both time domain signal mixing and direct signal fast transcoding. An exemplary embodiment of the present invention utilizes both time domain signal mixing and direct signal fast transcoding to process a bit-stream from a same channel during a conference.

    摘要翻译: 提供了一种利用时域信号混合和直接信号快速转码的会议系统。 本发明的示例性实施例利用时域信号混合和直接信号快速转码来在会议期间处理来自相同信道的比特流。