System and method for enhanced subjective stereo audio
    1.
    发明专利
    System and method for enhanced subjective stereo audio 有权
    用于增强主观立体声的系统和方法

    公开(公告)号:JP2009268137A

    公开(公告)日:2009-11-12

    申请号:JP2009157059

    申请日:2009-07-01

    CPC classification number: H04M9/082 H04S2420/05

    Abstract: PROBLEM TO BE SOLVED: To provide a system and a method for providing a stereo image with an optimal method for an echo canceller being combined by using a Haas effect to simulate sound picture providing better subjective impression of stereo sounds as compared to an objective stereo image, without compromise with subjective sensitivity of stereophone. SOLUTION: A far-end audio presenter unit includes a first adder configured to add L to R for creating a non-delayed monaural signal, a delay unit configured to delay the non-delayed monaural signal by prescribed time for creating a delayed monaural signal to be loaded to one or more main loudspeakers, and right and left loudspeakers each having one or more in number, to which a first and a second deduced L and/or R signals are loaded. COPYRIGHT: (C)2010,JPO&INPIT

    Abstract translation: 要解决的问题:提供一种用于提供立体声图像的系统和方法,其具有通过使用哈斯效应来组合的回声消除器的最佳方法,以模拟声音图像,从而提供立体声的更好的主观印象,与 客观的立体声图像,不影响立体声听觉的主观敏感性。 解决方案:远端音频呈现器单元包括:第一加法器,被配置为将L加到R以产生非延迟的单声道信号;延迟单元,被配置为将非延迟单声道信号延迟规定的时间,以产生延迟的 单声道信号被加载到一个或多个主扬声器以及每个具有一个或多个数字的左和右扬声器,第一和第二推导的L和/或R信号被加载到该扬声器。 版权所有(C)2010,JPO&INPIT

    Early detection of zeros in the transform domain
    2.
    发明申请
    Early detection of zeros in the transform domain 有权
    在变换域中早期检测零

    公开(公告)号:US20040264575A1

    公开(公告)日:2004-12-30

    申请号:US10831158

    申请日:2004-04-26

    CPC classification number: H04N19/14 H04N19/132 H04N19/176 H04N19/61

    Abstract: A method detects blocks that are to be indicated as skipped at an earlier stage of the encoding process, than would be the case with other implementations of the ITU H.263 and H.264 standards. The method includes transforming 4null4 blocks in the macro blocks having a skip vector of zero with a binary-transform function. The blocks having values of the four uppermost left binary-transform coefficients less than a predefined threshold, are defined as skipped, thus, minimizing the need for computationally demanding block transformation or quantization.

    Abstract translation: 一种方法检测在编码过程的较早阶段被跳过的块,与ITU H.263和H.264标准的其他实施方式相比。 该方法包括利用二进制变换函数来变换具有零跳跃向量的宏块中的4×4块。 将具有小于预定阈值的四个最上面的二进制变换系数的值的块定义为跳过,从而最小化对计算要求高的块变换或量化的需要。

    Echo canceller with reduced requirement for processing power
    3.
    发明申请
    Echo canceller with reduced requirement for processing power 有权
    回波消除器对处理能力要求降低

    公开(公告)号:US20040218755A1

    公开(公告)日:2004-11-04

    申请号:US10724043

    申请日:2003-12-01

    CPC classification number: H04M9/082

    Abstract: An echo canceller processing echo, noise and near end talk in a narrower, but still intelligible, frequency band for reducing required processing power and complexity. In a preferred embodiment of the present invention, an input audio signal of captured sound in an audio communication system is decimated and then divided into a number of sub bands by an analyze filter. Each sub band is processed as in background audio echo cancelling by subtracting the signal with an echo estimate from a model of the acoustic signal in the respective sub band, except from that the signal is also bypassed, adjusted by a filter and subtracted from the processed signal. The resulting signals are then recombined by a synthesize filter and interpolated to the original sampling rate and bandwidth. Finally, the output from the synthesize filter is added to the input audio signal, which has been delayed and adjusted by a filter. The filters are controlled by a control algorithm detecting the presence of near end sound, far end sound and noise, so that the filters, and consequently the high pass filter of the echo canceller, only pass high frequency (above low pass frequencies) when only near end sound is detected.

    Abstract translation: 回波消除器在较窄但仍然可理解的频带中处理回波,噪声和近端通话,以减少所需的处理能力和复杂性。 在本发明的优选实施例中,音频通信系统中的捕获声音的输入音频信号被抽取,然后由分析滤波器分成多个子频带。 通过从相应子带中的声学信号的模型中减去具有回波估计的信号来处理每个子带,除了信号被旁路之外,由滤波器调整并从处理的信号中减去 信号。 然后,所得到的信号由合成滤波器重新组合,并被内插到原始采样速率和带宽。 最后,将合成滤波器的输出添加到输入音频信号中,输入音频信号被滤波器延迟和调整。 滤波器由检测近端声音,远端声音和噪声的存在的控制算法控制,使得滤波器以及因此的回波消除器的高通滤波器仅在仅仅通过高频(高于低通频率)时才通过 检测到近端声音。

    Method for vector prediction
    4.
    发明申请
    Method for vector prediction 有权
    矢量预测方法

    公开(公告)号:US20040146110A1

    公开(公告)日:2004-07-29

    申请号:US10722479

    申请日:2003-11-28

    CPC classification number: H04N19/57 H04N19/52 H04N19/56

    Abstract: A method for prediction of the motion vector of a pixel block in a video picture that is to be coded. The actual motion vectors of two adjacent blocks close to the uppermost left corner of the block are selected as candidates for the prediction. One additional block, also adjacent to the block, is selected to decide which of the motion vectors to be used as the prediction. The vector difference to the motion vector of the decision block is decisive for the final selection.

    Method and apparatus for video compression
    7.
    发明申请
    Method and apparatus for video compression 有权
    视频压缩的方法和装置

    公开(公告)号:US20040233993A1

    公开(公告)日:2004-11-25

    申请号:US10844054

    申请日:2004-05-12

    CPC classification number: H04N19/85 H04N19/186 H04N19/593 H04N19/60

    Abstract: A unified solution to coding/decoding of different video formats such as 4:2:0, 4:2:2 and 4:4:4 is provided. A method of video coding includes transforming a first mnulln macro block of residual chrominance pixel values of moving pictures by a first integer-transform function generating a corresponding second mnulln macro block of integer-transform coefficients, further transforming DC values of the integer-transform coefficients by a second integer-transform function to generate a third block of integer-transformed DC coefficients. The method further includes generating the second mnulln macro block of integer-transform coefficients by utilizing a knullk integer-transform function on each knullk sub-block of the first mnulln macro block, wherein n and m are each a multiple of k, and generating the third block of coefficients by utilizing a second inullj integer-transform function on the DC values resulting in a (m/k)null(n/k) third block of integer-transformed DC coefficients.

    Abstract translation: 提供了对4:2:0,4:2:2和4:4:4等不同视频格式进行编码/解码的统一解决方案。 一种视频编码方法包括:通过产生对应的整数变换系数的第二m×m宏块的第一整数变换函数来变换运动图像的残余色度像素值的第一m×m宏块,进一步变换整数变换系数的DC值 通过第二整数变换函数来生成整数变换DC系数的第三块。 该方法还包括通过在第一m×n宏块的每个kxk子块上利用kxk整数变换函数来生成整数变换系数的第二m×n宏块,其中n和m分别为k的倍数,并且生成 通过利用第二个ixj整数变换函数,得到第二个整数变换的直流系数的第(m / k)×(n / k)个第三块的DC值的第三个系数块。

    Video teleconferencing system with digital transcoding

    公开(公告)号:US20030231600A1

    公开(公告)日:2003-12-18

    申请号:US10426245

    申请日:2003-04-29

    Inventor: Mark D. Polomski

    CPC classification number: H04N7/15 H04N7/152 H04N19/40

    Abstract: A video teleconferencing system uses digital transcoding to obtain algorithm transcoding, transmission rate matching, and spatial mixing. The video teleconferencing system comprises a multipoint control unit (MCU) for allowing multiple audiovisual terminals, which send and receive compressed digital data signals, to communicate with each other in a conference. The MCU has a video processing unit (VPU) that performs algorithm transcoding, rate matching, and spatial mixing among the terminals within a conference. The VPU includes a time division multiplex pixel bus and a plurality of processors. Each processor is assignable to an audiovisual terminal in the conference and is coupled to the pixel bus. In a receive mode, each processor receives and decodes compressed video signals from its assigned terminal and puts the decoded signal onto the pixel bus. In a transmit mode, each processor receives from the pixel bus uncompressed video signals from any terminal in the conference. The uncompressed video signals are processed and encoded for transmission to the respective assigned terminal. Video encoding time due to motion displacement search is reduced by passing displacement information from the compressed video signals to the encoder to be used directly or as a seed for further refinements of the motion displacement field.

    APPARATUS AND ASSOCIATED METHODOLOGY FOR SUPPRESSING AN ACOUSTIC ECHO
    9.
    发明申请
    APPARATUS AND ASSOCIATED METHODOLOGY FOR SUPPRESSING AN ACOUSTIC ECHO 审中-公开
    装置和相关方法用于抑制声学ECHO

    公开(公告)号:WO2011087376A1

    公开(公告)日:2011-07-21

    申请号:PCT/NO2011/000017

    申请日:2011-01-18

    CPC classification number: H04M9/085

    Abstract: A method for processing an audio signal executed by an audio echo suppression apparatus, the method including: receiving, at the audio echo suppression apparatus, the audio signal; generating a subband signal from the audio signal; delaying, at the audio echo suppression apparatus, the subband signal with a plurality of different delay values to form a plurality of time lag signals; multiplying, at the audio echo suppression apparatus, the plurality of time lag signals with first respective filter coefficients to generate a first signal; calculating, at the audio echo suppression apparatus, a complex product between pairs of the plurality of time lag signals to generate complex product signals; multiplying, at the audio echo suppression apparatus, each of a real part and imaginary part of the complex product signals with second respective filter coefficients, and taking a sum thereof, to generate a second signal; and estimating an echo subband signal from the first signal and the second signal.

    Abstract translation: 一种用于处理由音频回声抑制装置执行的音频信号的方法,所述方法包括:在音频回声抑制装置处接收音频信号; 从所述音频信号产生子带信号; 在音频回声抑制装置处延迟具有多个不同延迟值的子带信号以形成多个时滞信号; 在音频回声抑制装置处将具有第一相应滤波器系数的多个时滞信号相乘以产生第一信号; 在所述音频回声抑制装置处计算所述多个时滞信号对之间的复数乘积以产生复乘产品信号; 在音频回声抑制装置处,将具有第二相应滤波器系数的复乘积信号的实部和虚部中的每一者相乘并产生第二信号; 以及从所述第一信号和所述第二信号估计回波子带信号。

    METHOD AND SYSTEM FOR DETERMINING THE DIRECTION BETWEEN A DETECTION POINT AND AN ACOUSTIC SOURCE
    10.
    发明申请
    METHOD AND SYSTEM FOR DETERMINING THE DIRECTION BETWEEN A DETECTION POINT AND AN ACOUSTIC SOURCE 审中-公开
    用于确定检测点和声源之间的方向的方法和系统

    公开(公告)号:WO2011081527A1

    公开(公告)日:2011-07-07

    申请号:PCT/NO2010/000470

    申请日:2010-12-17

    Inventor: SOLVANG, Audun

    Abstract: A method and system for determining a direction between a detection point, e.g. at a camera in a video conference equipment, and an acoustic source, e.g. an active speaker participating in a video conference. The method comprises receiving acoustic signals originating from the acoustic source at a first and second pair of microphone elements, arranged symmetrically about the detection point; calculating a first cross correlation signal from the first pair of microphone elements; and calculating a second cross correlation of signals from the second pair of microphone elements. The direction is then calculated based on both the first and second cross correlation signals, e.g. by convolution. Further symmetrically arranged pairs of microphone elements may also be used.

    Abstract translation: 一种用于确定检测点之间的方向的方法和系统。 在视频会议设备中的相机处,以及声源,例如, 参与视频会议的主动演讲者。 该方法包括在第一和第二对麦克风元件处接收源自声源的声信号,围绕检测点对称地布置; 计算来自所述第一对麦克风元件的第一互相关信号; 以及计算来自所述第二对麦克风元件的信号的第二互相关。 然后基于第一和第二互相关信号,例如, 通过卷积 还可以使用对称布置的麦克风元件对。

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