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    • 31. 发明授权
    • Detection and suppression of flicker in a sequence of images
    • 检测和抑制图像序列中的闪烁
    • US08903171B2
    • 2014-12-02
    • US13521144
    • 2010-02-06
    • Bjorn VolckerWillem Bastiaan Kleijn
    • Bjorn VolckerWillem Bastiaan Kleijn
    • G06K9/00H04N5/235H04N7/01H04N5/14
    • H04N5/2357H04N5/144H04N7/0132
    • The invention relates to a method, device and computer-program product for detection of undesired temporal variations (flicker) in a sequence of video frames. In one embodiment, frame-wise luminance means are compared with a reference level and the crossing frequency is compared with expected variation frequencies, such as frequencies associated with an illumination frequency through aliasing. The crossings count can be refined by introducing a latency zone around the reference level. In case of a positive detection of an undesired temporal variation, there is further provided a correction method, device and computer-program product using cumulated distribution functions. The visual detriment of flicker-induced saturation of pixels is alleviated either by brightening non-saturated pixels or by replacing the saturated pixels by randomly sampled values in accordance with a reference cumulated distribution function. The invention provides embodiments suitable for real-time processing of streamed video sequences.
    • 本发明涉及用于检测视频帧序列中不需要的时间变化(闪烁)的方法,装置和计算机程序产品。 在一个实施例中,将逐帧亮度装置与参考电平进行比较,并将交叉频率与预期变化频率(例如通过混叠的照明频率相关联的频率)进行比较。 可以通过在参考级别周围引入延迟区域来改进交叉点计数。 在对不期望的时间变化进行肯定检测的情况下,还提供了使用累积分布函数的校正方法,设备和计算机程序产品。 通过增加非饱和像素或根据参考累积分布函数通过随机采样值替换饱和像素来减轻像素的闪烁引起的饱和度的视觉损害。 本发明提供适用于流式视频序列的实时处理的实施例。
    • 32. 发明授权
    • Method of coding a video signal
    • 编码视频信号的方法
    • US08582662B2
    • 2013-11-12
    • US13281087
    • 2011-10-25
    • Ermin KozicaDave ZachariahWillem Bastiaan Kleijn
    • Ermin KozicaDave ZachariahWillem Bastiaan Kleijn
    • H04N7/12
    • H04N19/136H04N19/12H04N19/172H04N19/44H04N19/46H04N19/61H04N21/23439H04N21/26275H04N21/4305
    • The invention relates to methods and apparatuses for encoding and decoding of a video sequence. In connection with encoding/decoding a video sequence it is desirable to increase the video quality without having to increase the bit-rate for the encoded video too much, thereby still providing a bit-efficient representation of the video. If multiple descriptions of the video sequence is used the invention improves the video quality without any increase of the bit-rate. According to the invention, this is achieved by using two or more coding units for encoding the same video sequence, wherein the encoding units perform their encoding operations displaced in time in relation to each other. Correspondingly, two or more decoding units are used for decoding the same video sequence, wherein the decoding units perform their decoding operations displaced in time in relation to each other.
    • 本发明涉及视频序列的编码和解码的方法和装置。 结合视频序列的编码/解码,期望增加视频质量,而不必增加编码视频的比特率太多,从而仍然提供视频的比特效率表示。 如果使用视频序列的多个描述,本发明可以在不增加比特率的情况下提高视频质量。 根据本发明,这通过使用两个或更多个编码单元来编码相同的视频序列来实现,其中编码单元执行它们相对于彼此在时间上移位的编码操作。 相应地,使用两个或多个解码单元来对相同的视频序列进行解码,其中解码单元相对于彼此在时间上执行其解码操作。
    • 33. 发明授权
    • Distributed blind source separation
    • 分布式盲源分离
    • US08423064B2
    • 2013-04-16
    • US13187998
    • 2011-07-21
    • Willem Bastiaan Kleijn
    • Willem Bastiaan Kleijn
    • H04B7/00H04B17/00H04B1/00H04B15/00
    • H04W8/005G10L21/0272H04L29/08H04L67/12H04W84/18
    • Systems and methods for using distributed processing in conjunction with blind source separation techniques for signal processing and acquisition in sensor network environments are provided. In the distributed blind source separation framework, sensors each perform some processing of sensor signals rather than transmitting such signals over long distances, and/or outside of the sensor network, to be processed at a central location. Sensors attempt to own a source signal, and a source signal can only be owned by one active sensor. Sensors that own a source signal broadcast the source signal directly or indirectly so that it is perceived by users. Sensors receive information from other sensors in their sensor neighborhood, including the observed signals of the other sensors and the estimated source signals of the sources owned by the other sensors. This allows all owning sensors to extract the respective source signals associated with the sources they own and all redundant sensors to check if there are any non-owned source signals present.
    • 提供了在传感器网络环境中使用分布式处理与盲源分离技术进行信号处理和采集的系统和方法。 在分布式盲源分离框架中,传感器各自执行传感器信号的一些处理,而不是在长距离和/或传感器网络外部传送这样的信号,以在中心位置处理。 传感器尝试拥有源信号,源信号只能由一个有源传感器拥有。 拥有源信号的传感器直接或间接地广播源信号,以便用户感知。 传感器从传感器附近的其他传感器接收信息,包括其他传感器的观测信号以及其他传感器所拥有的信号源的估计信号。 这允许所有拥有的传感器提取与其拥有的源相关联的各个源信号和所有冗余传感器,以检查是否存在任何非归属的源信号。
    • 34. 发明授权
    • Delay estimator
    • 延迟估计器
    • US08320552B2
    • 2012-11-27
    • US13139267
    • 2009-10-20
    • Willem Bastiaan Kleijn
    • Willem Bastiaan Kleijn
    • H04M1/00H04M9/00H04M9/08H04B3/20
    • H04M9/082H04B3/46
    • The present invention provides a method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the reference point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences.
    • 本发明提供了一种用于找到在两点之间行进的信号的延迟的估计的方法和装置。 在最后数量的时刻,在参考点和接收点处,从信号中估算出一个数量。 通过与适应于数量的典型量值的阈值进行比较来量化该值。 如果来自接收点的量化值相对于从参考点的量化值移回真实延迟,则量化值的某些共同出现具有非常低的概率。 因此,最佳延迟估计是产生最少数量的低概率共同出现的偏移。
    • 35. 发明申请
    • DETECTION AND SUPPRESSION OF FLICKER IN A SEQUENCE OF IMAGES
    • 检测和抑制图像序列中的闪烁
    • US20120281914A1
    • 2012-11-08
    • US13521144
    • 2010-02-06
    • Bjorn VolckerWillem Bastiaan Kleijn
    • Bjorn VolckerWillem Bastiaan Kleijn
    • G06K9/00
    • H04N5/2357H04N5/144H04N7/0132
    • The invention relates to a method, device and computer-program product for detection of undesired temporal variations (flicker) in a sequence of video frames. In one embodiment, frame-wise luminance means are compared with a reference level and the crossing frequency is compared with expected variation frequencies, such as frequencies associated with an illumination frequency through aliasing. The crossings count can be refined by introducing a latency zone around the reference level. In case of a positive detection of an undesired temporal variation, there is further provided a correction method, device and computer-program product using cumulated distribution functions. The visual detriment of flicker-induced saturation of pixels is alleviated either by brightening non-saturated pixels or by replacing the saturated pixels by randomly sampled values in accordance with a reference cumulated distribution function. The invention provides embodiments suitable for real-time processing of streamed video sequences.
    • 本发明涉及用于检测视频帧序列中不需要的时间变化(闪烁)的方法,装置和计算机程序产品。 在一个实施例中,将逐帧亮度装置与参考电平进行比较,并将交叉频率与预期变化频率(例如通过混叠的照明频率相关联的频率)进行比较。 可以通过在参考级别周围引入延迟区域来改进交叉点计数。 在对不期望的时间变化进行肯定检测的情况下,还提供了使用累积分布函数的校正方法,设备和计算机程序产品。 通过增加非饱和像素或根据参考累积分布函数通过随机采样值替换饱和像素来减轻像素的闪烁引起的饱和度的视觉损害。 本发明提供适用于流式视频序列的实时处理的实施例。
    • 36. 发明授权
    • Delay estimator
    • 延迟估计器
    • US08290143B2
    • 2012-10-16
    • US13245528
    • 2011-09-26
    • Willem Bastiaan Kleijn
    • Willem Bastiaan Kleijn
    • H04M9/08G10K11/16H03B29/00H04B3/20H04B1/38
    • H04M9/082H04B3/46
    • The present invention provides a method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the referenced point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences.
    • 本发明提供了一种用于找到在两点之间行进的信号的延迟的估计的方法和装置。 在最后数量的时刻,在参考点和接收点处,从信号中估算出一个数量。 通过与适应于数量的典型量值的阈值进行比较来量化该值。 如果来自接收点的量化值相对于从参考点的量化值移回真实延迟,则量化值的某些共同出现具有非常低的概率。 因此,最佳延迟估计是产生最少数量的低概率共同出现的偏移。
    • 37. 发明授权
    • Generalized analysis-by-synthesis speech coding method and apparatus
    • 广义综合语音编码方法和装置
    • US06169970A
    • 2001-01-02
    • US09004407
    • 1998-01-08
    • Willem Bastiaan Kleijn
    • Willem Bastiaan Kleijn
    • G10L1904
    • G10L19/12
    • A generalized analysis-by-synthesis method and apparatus are disclosed. A plurality of trial original signals are generated based on an original signal for coding. The trial original signals are constrained to be perceptually similar to the original signal. Trial original signals are coded to produce one or more parameters representative thereof. Estimates of the trial original signals are synthesized from these parameters. Errors between the trial original signals and the synthesized estimates are determined. A coded representation of the original signal is determined which comprises parameters of the trial original signal having an associated error which satisfies an error evaluation process. Trial original signals may be generated by application of time-warps or time-shifts to the original signal. Coding of a trial original signal may be performed with conventional analysis-by-synthesis coding such as code-excited linear prediction coding (CELP). A minimum square error process may serve as the error criterion.
    • 公开了一种综合的分析方法和装置。 基于用于编码的原始信号生成多个试用原始信号。 试用原始信号被限制在感觉上类似于原始信号。 试用原始信号被编码以产生代表其的一个或多个参数。 试验原始信号的估计是从这些参数合成的。 确定试用原始信号与合成估计之间的误差。 确定原始信号的编码表示,其包括具有满足错误评估处理的相关错误的试用原始信号的参数。 可以通过对原始信号施加时间经线或时移来产生试用原始信号。 可以用诸如代码激励线性预测编码(CELP)的常规综合编码来执行试用原始信号的编码。 最小平方误差过程可用作误差准则。
    • 38. 发明授权
    • Prototype waveform speech coding with interpolation of pitch,
pitch-period waveforms, and synthesis filter
    • 具有间距,音调周期波形和合成滤波器插值的原型波形语音编码
    • US5884253A
    • 1999-03-16
    • US943329
    • 1997-10-03
    • Willem Bastiaan Kleijn
    • Willem Bastiaan Kleijn
    • G10L19/08G10L5/02
    • G10L19/097
    • A speech coding system providing reconstructed voiced speech with a smoothly evolving pitch-cycle waveform. A speech signal is represented by isolating and coding prototype waveforms. Each prototype waveform is an exemplary pitch-cycle of voiced speech. A coded prototype waveform is transmitted at regular intervals to a receiver which synthesizes (or reconstructs) an estimate of the original speech segment based on the prototypes. The estimate of the original speech signal is provided by a prototype interpolation process which provides a smooth time-evolution of pitch-cycle waveforms in the reconstructed speech. Illustratively, a frame of original speech is coded by first filtering the frame with a linear predictive filter. Next a pitch-cycle of the filtered original is identified and extracted as a prototype waveform. The prototype waveform is then represented as a set of Fourier series (frequency domain) coefficients. The pitch-period and Fourier coefficients of the prototype, as well as the parameters of the linear predictive filter, are used to represent a frame of original speech. These parameters are coded by vector and scalar quantization and communicated over a channel to a receiver which uses information representing two consecutive frames to reconstruct the earlier of the two frames based on a continuous prototype waveform interpolation process. Waveform interpolation may be combined with conventional CELP techniques for coding unvoiced portions of the original speech signal.
    • 一种语音编码系统,其提供具有平滑演进的音调周期波形的重构语音。 通过分离和编码原型波形来表示语音信号。 每个原型波形是有声语音的示例性音调周期。 经编码的原型波形以规则的间隔传输到基于原型合成(或重构)原始语音段的估计的接收机。 原始语音信号的估计是通过原型内插处理来提供的,该原理插值处理在重构语音中提供音调周期波形的平滑时间演变。 说明性地,通过用线性预测滤波器首先滤波帧来对原始语音的帧进行编码。 接下来,将经滤波的原稿的音调周期识别并提取为原型波形。 原型波形然后被表示为一组傅里叶级数(频域)系数。 原型的音调周期和傅立叶系数以及线性预测滤波器的参数用于表示原始语音的帧。 这些参数通过向量和标量量化进行编码,并通过信道传送到接收机,接收机使用表示两个连续帧的信息,以基于连续的原型波形插值处理来重建两个帧中较早的帧。 波形内插可以与用于编码原始语音信号的无声部分的常规CELP技术组合。
    • 39. 发明授权
    • Speech coding parameter sequence reconstruction by sequence
classification and interpolation
    • 通过序列分类和插值的语音编码参数序列重建
    • US5839102A
    • 1998-11-17
    • US346798
    • 1994-11-30
    • Jesper HaagenWillem Bastiaan Kleijn
    • Jesper HaagenWillem Bastiaan Kleijn
    • G10L13/00G10L19/00G10L9/18
    • G10L19/0018G10L25/51
    • A method and apparatus which allows the transmission of the perceptually important features of a speech-coding parameter at a low bit rate. The speech coding parameter may, for example, comprise the signal power of the speech. The parameter is processed on a block by block basis. The parameter value at the block boundaries is transmitted by conventional methods such as, for example, by means of differential quantization. The shape of the reconstructed parameter contour within block boundaries is based on a classification. The classification determines perceptually important features of the parameter contour within a block. The classification can be performed either at the transmitting end of the coder (using, for example, the original parameter contour with high time resolution and possibly other speech parameters as well) or at the receiving end of the coder (using, for example, the transmitted parameter values, and possibly other transmitted speech parameters as well). Based on the result of the classification as well as the parameter values at the block boundaries, a parameter contour (within the block) is selected from an inventory of possible parameter contours. The inventory may include a linear interpolation contour and a step function contour. The step function contour may be particularly useful when the features indicate the presence of a plosive. The inventory may adapt to the transmitted parameter values at the block boundaries.
    • 允许以低比特率传输语音编码参数的感知重要特征的方法和装置。 语音编码参数可以例如包括语音的信号功率。 该参数以块为单位进行处理。 块边界处的参数值通过常规方法传输,例如通过差分量化。 块边界内的重构参数轮廓的形状基于分类。 分类确定块内参数轮廓的感知重要特征。 该分类可以在编码器的发送端执行(例如,使用例如具有高时间分辨率的原始参数轮廓以及可能的其他语音参数)或在编码器的接收端(例如使用 传输的参数值,以及可能的其他传输的语音参数)。 根据分类结果以及块边界处的参数值,从可能的参数轮廓的库存中选择参数轮廓(在块内)。 库存可以包括线性内插轮廓和阶梯函数轮廓。 当功能指示爆破的存在时,阶梯功能轮廓可能特别有用。 库存可以适应块边界处的传输参数值。
    • 40. 发明授权
    • RCELP coder
    • RCELP编码器
    • US5704003A
    • 1997-12-30
    • US530040
    • 1995-09-19
    • Willem Bastiaan KleijnDror Nahumi
    • Willem Bastiaan KleijnDror Nahumi
    • G10L19/04G10L19/00G10L19/08G10L19/12H03M7/30G10L5/00
    • G10L19/09
    • An improved method of speech coding for use in conjunction with speech coding methods wherein speech is digitized into a plurality of temporally defined frames, each frame including a plurality of sub-frames, and the digitized speech is partitioned into periodic components and a residual signal. For each of a plurality of sub-frames of the residual signal, the improved method of speech coding selects and applies a time shift T to the sub-frame by applying a matching criterion to (a) the current sub-frame of the residual signal, and (b) a sample-to-sample (subframe-to-subframe) pitch delay determined by applying linear interpolation to known pitch delays occurring at or near frame-to-frame boundaries of previous frames. The matching criterion is applied by minimizing .epsilon., where: ##EQU1## (r(n-T)) is the residual signal of the current frame shifted by time T, r(n-D(n)) is the delayed residual signal from a previously-occurring frame, n is a positive integer, r is the instantaneous amplitude of the residual signal, and D(n) is the sample-to-sample pitch delay determined by applying linear interpolation to known pitch delay values occurring at or near frame-to-frame boundaries.
    • 一种用于语音编码方法的改进的语音编码方法,其中语音被数字化为多个时间上定义的帧,每个帧包括多个子帧,并且数字化语音被划分为周期分量和残差信号。 对于剩余信号的多个子帧中的每一个,改进的语音编码方法通过对(a)剩余信号的当前子帧应用匹配准则来选择并应用时移T到子帧 ,以及(b)通过对在先前帧的帧到帧边界处或附近发生的已知音调延迟应用线性内插而确定的采样到采样(子帧到子帧)的音调延迟。 通过最小化ε来应用匹配标准,其中: