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1. 发明申请

WO2011107434A1 DISTRIBUTION-CONSTRAINED QUANTIZATION 审中-公开
标题翻译：分布约束量化
公开(公告)号：WO2011107434A1
公开(公告)日：2011-09-09
申请号：PCT/EP2011/052936
申请日：2011-02-28
申请人： GLOBAL IP SOLUTIONS (GIPS) AB , GLOBAL IP SOLUTIONS, INC. , LI, Minyue , KLEIJN, Willem, Bastiaan
发明人： LI, Minyue , KLEIJN, Willem, Bastiaan
IPC分类号： G10L19/00 , H04N7/26
CPC分类号： G10L19/00 , G10L19/035 , H04N19/126 , H04N19/147 , H04N19/60 , H04N19/94
摘要： The invention relates to improved signal quantization and inter alia addresses the problem of reducing artefacts in the reconstructed signal that arise from unintended statistical correlation between quantization errors and signal values. In one aspect, the invention provides an encoder characterized by the fact that several quantizers (determined by a partition of quantization cells and reconstruction points) are applied, one at a time, in an alternating fashion. In another aspect, the invention provides a decoder characterized by a corresponding alternation of quantizers, so that quantized samples are reconstructed by a corresponding quantizer on the decoder side. The invention further provides a method of manufacturing an encoder, decoder or quantizer according to the invention, which includes designing an ensemble of quantizers.
摘要翻译：本发明涉及改进的信号量化，并且尤其涉及减少由量化误差和信号值之间的非预期的统计相关引起的重构信号中的伪像的问题。一方面，本发明提供了一种编码器，其特征在于以交替的方式一个一个地施加几个量化器（由量化单元和重构点的分区确定）。在另一方面，本发明提供了一种解码器，其特征在于相应的量化器交替，使得量化样本由解码器侧的相应量化器重构。本发明还提供了一种制造根据本发明的编码器，解码器或量化器的方法，其包括设计量化器的集合。

2. 发明申请

WO2010066496A1 DELAY ESTIMATOR 审中-公开
标题翻译：延迟估计
公开(公告)号：WO2010066496A1
公开(公告)日：2010-06-17
申请号：PCT/EP2009/063707
申请日：2009-10-20
申请人： GLOBAL IP SOLUTIONS (GIPS) AB , GLOBAL IP SOLUTIONS, INC. , KLEIJN, Willem Bastiaan
发明人： KLEIJN, Willem Bastiaan
IPC分类号： H04M9/08
CPC分类号： H04M9/082 , H04B3/46
摘要： 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.
摘要翻译：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。

3. 发明申请

WO2009000821A1 METHOD AND RECEIVER FOR DETERMINING A JITTER BUFFER LEVEL 审中-公开
标题翻译：用于确定抖动缓冲器电平的方法和接收器
公开(公告)号：WO2009000821A1
公开(公告)日：2008-12-31
申请号：PCT/EP2008/057991
申请日：2008-06-24
申请人： Global IP Solutions (GIPS) AB , Global IP Solutions Inc. , FAHLBERG LUNDIN, Henrik
发明人： FAHLBERG LUNDIN, Henrik
IPC分类号： H04J3/06 , H04L12/64
CPC分类号： H04J3/0632 , H04L47/30
摘要： The invention relates to a method and a receiver having control logic means for determining a target packet level of a jitter buffer adapted to receive packets with digitized signal samples, which packets are subject to delay jitter, from a packet data network. According to the invention, the jitter buffer is made adaptive to current network conditions, i.e., the nature and magnitude of the jitter observed by the receiver, by collecting statistical measures that describe these conditions. The target buffer level is determined with regard to the effect of packet losses in terms of duration of the discontinued playback of the true signal. This effect is derived from statistical measures of the network conditions as perceived by the receiving side and as reflected by a probability mass function which is continuously updated with packet inter-arrival times. The target buffer level is the result of minimization of a cost function which weights the internal buffer delay and an expected length of buffer underflow.
摘要翻译：本发明涉及一种具有控制逻辑装置的控制逻辑装置，该方法和接收机具有用于确定抖动缓冲器的目标分组级别，该抖动缓冲器适用于从分组数据网络接收具有数字化信号样本（哪个分组遭受延迟抖动）的分组。根据本发明，通过收集描述这些条件的统计测量，使抖动缓冲器适应于当前网络条件，即接收机观察到的抖动的性质和幅度。关于分组丢失对真实信号的停止重放的持续时间的影响来确定目标缓冲器电平。这种效应来源于接收侧所感知的网络条件的统计测量，并且由具有分组到达时间连续更新的概率质量函数反映出来。目标缓冲器电平是对内部缓冲器延迟和缓冲器下溢的预期长度进行加权的成本函数最小化的结果。

4. 发明申请

WO2009015944A1 A LOW-DELAY AUDIO CODER 审中-公开
标题翻译：低延迟音频编码器
公开(公告)号：WO2009015944A1
公开(公告)日：2009-02-05
申请号：PCT/EP2008/057970
申请日：2008-06-23
申请人： Global IP Solutions (GIPS) AB , Global IP Solutions, Inc. , LI, Minyue , KLEIJN, Willem Bastiaan
发明人： LI, Minyue , KLEIJN, Willem Bastiaan
IPC分类号： G10L19/00
CPC分类号： G10L19/0017 , H04R25/552 , H04R2225/55
摘要： The present invention relates to methods and devices for encoding and decoding digital audio signals, e.g. a speech signal. An audio coder and a decoder are provided wherein a modeller adds a first distribution model obtained from model parameters of past segments of the digital audio signal and a fixed distribution model, each of the models being multiplied by a weighting coefficient, for obtaining a combined distribution model. The weighting coefficients are selected to minimize a code length of a current segment of the digital audio signal. As the combined distribution model is a sum of several distribution models, wherein at least some of the models is based on the model parameters, flexibility is introduced in the signal model used to encode the digital audio signal. Thus, an audio coder and decoder providing a low bit rate in average, low bit rate variations and low error propagation are provided.
摘要翻译：本发明涉及用于对数字音频信号进行编码和解码的方法和装置，例如，一个语音信号。提供一种音频编码器和解码器，其中，建模器添加从数字音频信号的过去段的模型参数和固定分布模型获得的第一分布模型，每个模型乘以加权系数，以获得组合分布模型。选择加权系数以最小化数字音频信号的当前段的代码长度。由于组合分布模型是几个分布模型的总和，其中至少一些模型基于模型参数，在用于对数字音频信号进行编码的信号模型中引入了灵活性。因此，提供了平均提供低比特率，低比特率变化和低错误传播的音频编码器和解码器。

5. 发明申请

WO2007131628A1 AUDIO MIXING 审中-公开
标题翻译：音频混音
公开(公告)号：WO2007131628A1
公开(公告)日：2007-11-22
申请号：PCT/EP2007/003844
申请日：2007-05-02
申请人： GLOBAL IP SOLUTIONS (GIPS) AB , GLOBAL IP SOLUTIONS, INC. , ENBOM, Niklas , HAGEN, Roar
发明人： ENBOM, Niklas , HAGEN, Roar
IPC分类号： H04M3/56 , H04R3/00
CPC分类号： H04R3/005 , H04M3/563 , H04M3/569 , H04M2201/18
摘要： The invention relates to a method and an apparatus for mixing audio signals. The invention is based on the idea to base selection of what audio channels to mix on the dynamic behaviour of the audio channels in terms of whether they are active or not, rather than basing the selection on quantative measures of the audio channels. According to the invention, a set of audio channels are arranged in a mixing stack and a mixed audio signal is produced from the topmost channels above a predetermined threshold level in the stack. Whenever a channel becomes active, it is placed at the top of the stack.
摘要翻译：本发明涉及一种用于混合音频信号的方法和装置。本发明基于这样的想法，即基于音频信道的动态行为来选择什么音频信道，以便它们是否是活动的，而不是基于对音频信道的量化度量的选择。根据本发明，一组音频通道被布置在混合堆栈中，并且混合音频信号从堆叠中的预定阈值电平以上的最上面的信道产生。每当一个通道变为活动状态时，它将被放置在堆栈的顶部。

6. 发明申请

WO2009156280A1 VIDEO SELECTOR 审中-公开
标题翻译：视频选择器
公开(公告)号：WO2009156280A1
公开(公告)日：2009-12-30
申请号：PCT/EP2009/057252
申请日：2009-06-11
申请人： GLOBAL IP SOLUTIONS (GIPS) AB , Global IP Solutions, Inc. , ELLNER, Henrik
发明人： ELLNER, Henrik
IPC分类号： H04N7/15
CPC分类号： H04N7/15 , H04N7/147 , H04N7/152
摘要： The invention provides a method for selecting input video signals in a video conference for transmission. The method bases its selection decision on the momentary voice activity of the participants, but does so while observing constraints aimed at: • improving stability of the image seen by each participant; • decreasing flickering of the image; • choosing comfortable and helpful swapping times between different image configurations; and • reflecting the importance which is attributed to persons having a higher past average activity in a natural conference. In one embodiment of the invention, the set of constraints includes an override, by which one or more participants can be selected irrespectively of the selection decision. In accordance with a second aspect of the invention, there is provided video selector hardware comprising units to perform the steps of the method.
摘要翻译：本发明提供一种在视频会议中选择输入视频信号以进行传输的方法。该方法基于对参与者的瞬间语音活动的选择决定，但是在遵守旨在：提高每个参与者看到的图像的稳定性的约束下进行选择。减少图像的闪烁; 选择不同图像配置之间舒适有用的交换时间; 反映了自然会议过去平均活动人数的重要性。在本发明的一个实施例中，所述约束集包括覆盖，通过该覆盖可以选择一个或多个参与者而不考虑选择决定。根据本发明的第二方面，提供了包括用于执行该方法的步骤的单元的视频选择器硬件。

7. 发明申请

WO2011036123A1 METHOD AND DEVICE FOR DETERMINING A JITTER BUFFER LEVEL 审中-公开
标题翻译：用于确定抖动缓冲器电平的方法和装置
公开(公告)号：WO2011036123A1
公开(公告)日：2011-03-31
申请号：PCT/EP2010/063818
申请日：2010-09-20
申请人： Global IP Solutions (GIPS) AB , Global IP Solutions, Inc. , FAHLBERG LUNDIN, Henrik , HOLMER, Stefan
发明人： FAHLBERG LUNDIN, Henrik , HOLMER, Stefan
IPC分类号： H04L12/56 , H04L12/26
CPC分类号： H04L43/087 , H04L43/0852 , H04L43/106 , H04L49/90 , H04L49/9005
摘要： It is disclosed a method for determining a buffer level of a jitter data buffer (203) comprised in a receiver (200) adapted to sequentially receive data packets from a communications network (202), wherein frames are encoded into the data packets. Each frame may comprise timestamp information and payload size information. The buffer level is determined on the basis of a first part of frame arrival delay related to payload size variation between frames and a second part related to the amount of crosstraffic in the communications network (202). It is further disclosed a receiver (200) adapted to sequentially receive data packets from a communications network (202), wherein frames are encoded into the data packets, the receiver (200) comprising a data jitter buffer (203) and being adapted to perform the method.
摘要翻译：公开了一种用于确定包括在接收器（200）中的抖动数据缓冲器（203）的缓冲器电平的方法，该接收机适于顺序地从通信网络（202）接收数据分组，其中帧被编码到数据分组中。每个帧可以包括时间戳信息和有效载荷大小信息。基于与帧之间的有效负载大小变化相关的帧到达延迟的第一部分和与通信网络（202）中的交叉量相关的第二部分，确定缓冲器电平。还公开了一种适于从通信网络（202）顺序地接收数据分组的接收机（200），其中帧被编码为数据分组，所述接收机（200）包括数据抖动缓冲器（203）并且适于执行方法。

8. 发明申请

WO2011033103A1 CODING AND DECODING OF SOURCE SIGNALS USING CONSTRAINED RELATIVE ENTROPY QUANTIZATION 审中-公开
标题翻译：使用约束相关熵定标对源信号进行编码和解码
公开(公告)号：WO2011033103A1
公开(公告)日：2011-03-24
申请号：PCT/EP2010/063789
申请日：2010-09-20
申请人： GLOBAL IP SOLUTIONS (GIPS) AB , GLOBAL IP SOLUTIONS, INC. , KLEIJN, Willem Bastiaan , LI, Minyue
发明人： KLEIJN, Willem Bastiaan , LI, Minyue
IPC分类号： G10L19/00 , H03M7/30 , H04N7/26
CPC分类号： G10L19/00
摘要： Methods and devices for encoding and decoding are provided. A source signal value is encoded by a quantization index determined using a partition into quantization cells. Decoding of the quantization index takes place by sampling a reconstruction probability distribution, thereby obtaining a reconstructed signal value, such that the reconstructed signal value lies in the same quantization cell as the source signal value. In one embodiment, encoding and decoding are such that their succession preserves the source signal distribution. In another embodiment, the partition and the reconstruction probability distribution are determined in such manner that the quantization error is minimized subject to a constraint on the relative entropy between the source signal and the reconstructed signal.
摘要翻译：提供了用于编码和解码的方法和装置。源信号值由使用分区确定的量化单元的量化索引进行编码。通过对重构概率分布进行采样来进行量化索引的解码，从而获得重构信号值，使得重构信号值位于与源信号值相同的量化单元中。在一个实施例中，编码和解码使得它们的继承保持源信号分布。在另一个实施例中，以这样的方式确定分区和重建概率分布，使得在对源信号和重建信号之间的相对熵的约束下，量化误差被最小化。

9. 发明申请

WO2010142591A1 DETECTION AND SUPPRESSION OF FLICKER IN A SEQUENCE OF IMAGES 审中-公开
标题翻译：按图像序列检测和抑制闪烁
公开(公告)号：WO2010142591A1
公开(公告)日：2010-12-16
申请号：PCT/EP2010/057752
申请日：2010-06-02
申请人： GLOBAL IP SOLUTIONS (GIPS) AB , GLOBAL IP SOLUTIONS, INC. , VÖLCKER, Björn , KLEIJN, Willem, Bastiaan
发明人： VÖLCKER, Björn , KLEIJN, Willem, Bastiaan
IPC分类号： H04N5/44 , H04N5/235
CPC分类号： H04N5/2357 , H04N5/144 , H04N7/0132
摘要： 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.
摘要翻译：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，设备和计算机程序产品。在一个实施例中，将逐帧亮度装置与参考电平进行比较，并将交叉频率与期望的变化频率（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。在不期望的时间变化的正检测的情况下，还提供了使用累积分布函数的校正方法，装置和计算机程序产品。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上的损害。本发明提供适用于流式视频序列的实时处理的实施例。

10. 发明申请

WO2008125523A1 ADAPTIVE, SCALABLE PACKET LOSS RECOVERY 审中-公开
标题翻译：自适应，可分级包丢失恢复
公开(公告)号：WO2008125523A1
公开(公告)日：2008-10-23
申请号：PCT/EP2008/054114
申请日：2008-04-04
申请人： Global IP Solutions (GIPS) AB , Global IP Solutions, Inc. , FELDBAUER, Christian , KLEIJN, Willem Bastiaan
发明人： FELDBAUER, Christian , KLEIJN, Willem Bastiaan
IPC分类号： H03M7/30 , H03M13/37
CPC分类号： G06F11/10 , G10L19/005 , G10L19/24 , H03M7/30 , H03M13/373 , H03M13/6312 , H04N19/115 , H04N19/15 , H04N19/164 , H04N19/166
摘要： A system for transmitting data packets representing a source signal across a packet data network is provided. The encoder comprises a first encoder (110) and a redundancy encoder (120). The redundancy encoding is generated with a bit rate continuously scalable, the bit rate being provided by a bit rate controller (142) that uses input from the network (130) and packet-loss rate information. At the decoder, recovery is performed by a parameter estimator based in part on information transmitted from the first encoder using information from previous and/or future blocks and in addition on redundant information. The method may be added to existing lossy source coding systems or may be used to enhance the quality of the reconstructed source signal even in scenarios without packet loss.
摘要翻译：提供了一种用于在分组数据网络上传送表示源信号的数据分组的系统。编码器包括第一编码器（110）和冗余编码器（120）。以比特率连续可缩放地生成冗余编码，比特率由使用来自网络（130）的输入的比特率控制器（142）和分组丢失率信息提供。在解码器处，部分地基于来自第一编码器的信息，使用来自先前和/或未来块的信息以及另外的冗余信息，由参数估计器执行恢复。该方法可以添加到现有的有损耗源编码系统中，或者即使在没有分组丢失的情况下也可以用于增强重构源信号的质量。

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