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1.发明申请Bit rate reduction in audio encoders by exploiting inharmonicity effects and auditory temporal masking 审中-公开音频编码器通过利用不协调效应和听觉时间屏蔽来降低比特率公开(公告)号：US20080221875A1
公开(公告)日：2008-09-11
申请号：US12153408
申请日：2008-05-19
CPC分类号： G10L19/02 , G10L19/032
摘要： The present invention relates to a method for encoding an audio signal. In a first embodiment a model relating to temporal masking of sound provided to a human ear is provided. A temporal masking index is determined in dependence upon a received audio signal and the model using a forward and a backward masking function. Using a psychoacoustic model a masking threshold is determined in dependence upon the temporal masking index. Finally, the audio signal is encoded in dependence upon the masking threshold. The method has been implemented using the MPEG-1 psychoacoustic model 2. Semiformal listening test showed that using the method for encoding an audio signal according to the present invention the subjective high quality of the decoded compressed sounds has been maintained while the bit rate was reduced by approximately 10%. In a second embodiment, the inharmonic structure of audio signals is modeled and incorporated into the MPEG-1 psychoacoustic model 2. In the model, the relationship between the spectral components of the input audio signal is considered and an inharmonicity index is defined and incorporated into the MPEG-1 psychoacoustic model 2. Informal listening tests have shown that the bit rate required for transparent coding of inharmonic (multi-tonal) audio material can be reduced by 10% if the modified psychoacoustic model 2 is used in the MPEG 1 Layer II encoder.
摘要翻译： 本发明涉及一种对音频信号进行编码的方法。 在第一实施例中，提供了与提供给人耳的声音的时间屏蔽有关的模型。 根据接收的音频信号和使用前向和后向屏蔽功能的模型来确定时间屏蔽索引。 使用心理声学模型，根据时间掩蔽指数确定掩蔽阈值。 最后，根据屏蔽阈值对音频信号进行编码。 该方法已经使用MPEG-1心理声学模型2实现。 半形式听力测试表明，使用根据本发明的音频信号编码方法，解码压缩声音的主观高品质已被维持，同时比特率降低了约10％。 在第二实施例中，音频信号的非谐结构被建模并且并入到MPEG-1心理声学模型2中。 在该模型中，考虑了输入音频信号的频谱分量之间的关系，并定义了非均匀性指标并将其并入到MPEG-1心理声学模型2中。 非正式听力测试表明，如果在MPEG 1 Layer II编码器中使用经修改的心理声学模型2，则对于非音调（多音调）音频材料的透明编码所需的比特率可以降低10％。
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2.发明授权Bit rate reduction in audio encoders by exploiting inharmonicity effects and auditory temporal masking 失效音频编码器通过利用不协调效应和听觉时间屏蔽来降低比特率公开(公告)号：US07398204B2
公开(公告)日：2008-07-08
申请号：US10647320
申请日：2003-08-26
CPC分类号： G10L19/02 , G10L19/032
摘要： The present invention relates to a method for encoding an audio signal. In a first embodiment a model relating to temporal masking of sound provided to a human ear is provided. A temporal masking index is determined in dependence upon a received audio signal and the model using a forward and a backward masking function. Using a psychoacoustic model a masking threshold is determined in dependence upon the temporal masking index. Finally, the audio signal is encoded in dependence upon the masking threshold. The method has been implemented using the MPEG-1 psychoacoustic model 2. Semiformal listening test showed that using the method for encoding an audio signal according to the present invention the subjective high quality of the decoded compressed sounds has been maintained while the bit rate was reduced by approximately 10%. In a second embodiment, the inharmonic structure of audio signals is modeled and incorporated into the MPEG-1 psychoacoustic model 2. In the model, the relationship between the spectral components of the input audio signal is considered and an inharmonicity index is defined and incorporated into the MPEG-1 psychoacoustic model 2. Informal listening tests have shown that the bit rate required for transparent coding of inharmonic (multi-tonal) audio material can be reduced by 10% if the modified psychoacoustic model 2 is used in the MPEG 1 Layer II encoder.
摘要翻译： 本发明涉及一种对音频信号进行编码的方法。 在第一实施例中，提供了与提供给人耳的声音的时间屏蔽有关的模型。 根据接收的音频信号和使用前向和后向屏蔽功能的模型来确定时间屏蔽索引。 使用心理声学模型，根据时间掩蔽指数确定掩蔽阈值。 最后，根据屏蔽阈值对音频信号进行编码。 该方法已经使用MPEG-1心理声学模型实现2.半形式听力测试表明，使用根据本发明的音频信号编码方法，解码的压缩声音的主观高质量已被维持，同时比特率降低 减少约10％。 在第二实施例中，音频信号的非谐结构被建模并且并入到MPEG-1心理声学模型2中。在该模型中，考虑了输入音频信号的频谱分量之间的关系，并且将不协调性指数定义并并入 MPEG-1心理声学模型2.非正式听力测试表明，如果在MPEG 1 Layer II中使用修改的心理声学模型2，则可以将非音调（多音调）音频素材的透明编码所需的比特率降低10％ 编码器。
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公开(公告)号：US08711015B2
公开(公告)日：2014-04-29
申请号：US13216295
申请日：2011-08-24
申请人： Frederic Mustiere , Hossein Najaf-Zadeh , Ramin Pishehvar , Hassan Lahdili , Louis Thibault , Martin Bouchard
发明人： Frederic Mustiere , Hossein Najaf-Zadeh , Ramin Pishehvar , Hassan Lahdili , Louis Thibault , Martin Bouchard
IPC分类号： H03M7/30
CPC分类号： H03M7/30
摘要： The invention relates to compressing of sparse data sets contains sequences of data values and position information therefor. The position information may be in the form of position indices defining active positions of the data values in a sparse vector of length N. The position information is encoded into the data values by adjusting one or more of the data values within a pre-defined tolerance range, so that a pre-defined mapping function of the data values and their positions is close to a target value. In one embodiment, the mapping function is defined using a sub-set of N filler values which elements are used to fill empty positions in the input sparse data vector. At the decoder, the correct data positions are identified by searching though possible sub-sets of filler values.
摘要翻译： 本发明涉及压缩稀疏数据集包含数据值序列和位置信息。 位置信息可以是以长度N的稀疏向量定义数据值的活动位置的位置索引的形式。通过将一个或多个数据值调整在预定公差内，将位置信息编码为数据值 范围，使得数据值及其位置的预定义映射函数接近目标值。 在一个实施例中，使用N个填充值的子集定义映射函数，该子集用于填充输入稀疏数据向量中的空白位置。 在解码器处，通过搜索填充值可能的子集来识别正确的数据位置。
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公开(公告)号：US20130253923A1
公开(公告)日：2013-09-26
申请号：US13426217
申请日：2012-03-21
申请人： Frederic Mustiere , Martin Bouchard , Hossein Najaf-Zadeh , Louis Thibault , Raman Pishehvar , Hassan Lahdili
发明人： Frederic Mustiere , Martin Bouchard , Hossein Najaf-Zadeh , Louis Thibault , Raman Pishehvar , Hassan Lahdili
CPC分类号： G10L21/0216 , G10L2021/02166 , H04R3/005
摘要： A method is disclosed for maintaining spatial queues in digital sound signals. Sound signals are received from each of a plurality of transducers. The sound signals are transformed using a common real-valued spectral gain, G, to maintain spatial cues within the sound signals, the common spectral gain, G, determined by: calculating G as a function of a derivative of a known cost function and as a function of at least one multichannel frequency-domain Bayesian short-time estimator.
摘要翻译： 公开了一种用于维持数字声音信号中的空间队列的方法。 从多个换能器中的每一个接收声音信号。 声音信号使用公共实数频谱增益G进行变换，以保持声音信号内的空间提示，公共频谱增益G由下式确定：将G作为已知成本函数的导数和 至少一个多通道频域贝叶斯短时间估计器的功能。
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公开(公告)号：US20120053948A1
公开(公告)日：2012-03-01
申请号：US13216295
申请日：2011-08-24
申请人： Frederic Mustiere , Hossein Najaf-Zadeh , Ramin Pishehvar , Hassan Lahdili , Louis Thibault , Martin Bouchard
发明人： Frederic Mustiere , Hossein Najaf-Zadeh , Ramin Pishehvar , Hassan Lahdili , Louis Thibault , Martin Bouchard
CPC分类号： H03M7/30
摘要： The invention relates to compressing of sparse data sets contains sequences of data values and position information therefor. The position information may be in the form of position indices defining active positions of the data values in a sparse vector of length N. The position information is encoded into the data values by adjusting one or more of the data values within a pre-defined tolerance range, so that a pre-defined mapping function of the data values and their positions is close to a target value. In one embodiment, the mapping function is defined using a sub-set of N filler values which elements are used to fill empty positions in the input sparse data vector. At the decoder, the correct data positions are identified by searching though possible sub-sets of filler values.
摘要翻译： 本发明涉及压缩稀疏数据集包含数据值序列和位置信息。 位置信息可以是以长度N的稀疏向量定义数据值的活动位置的位置索引的形式。通过将一个或多个数据值调整在预定公差内，将位置信息编码为数据值 范围，使得数据值及其位置的预定义映射函数接近目标值。 在一个实施例中，使用N个填充值的子集定义映射函数，该子集用于填充输入稀疏数据向量中的空白位置。 在解码器处，通过搜索填充值可能的子集来识别正确的数据位置。
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公开(公告)号：US20120023051A1
公开(公告)日：2012-01-26
申请号：US13188915
申请日：2011-07-22
申请人： Ramin PISHEHVAR , Christopher Srinivasa , Hossein Najaf-Zadeh , Frederic Mustiere , Hassan Lahdili , Louis Thibault
发明人： Ramin PISHEHVAR , Christopher Srinivasa , Hossein Najaf-Zadeh , Frederic Mustiere , Hassan Lahdili , Louis Thibault
IPC分类号： G06N3/02
CPC分类号： G06N3/049
摘要： The invention relates to sparse parallel signal coding using a neural network which parameters are adaptively determined in dependence on a pre-determined signal shaping characteristic. A signal is provides to a neural network encoder implementing a locally competitive algorithm for sparsely representing the signal. A plurality of interconnected nodes receive projections of the input signal, and each node generates an output once an internal potential thereof exceeds a node-dependent threshold value. The node-dependent threshold value for each of the nodes is set based upon the pre-determined shaping characteristic. In one embodiment, the invention enables to incorporate perceptual auditory masking in the sparse parallel coding of audio signals.
摘要翻译： 本发明涉及使用神经网络的稀疏并行信号编码，这些参数根据预定信号整形特性自适应地确定。 信号提供给实现用于稀疏地表示信号的局部竞争算法的神经网络编码器。 多个相互连接的节点接收输入信号的投影，并且一旦其内部电位超过节点相关阈值，每个节点就产生输出。 基于预定的整形特性来设置每个节点的节点依赖阈值。 在一个实施例中，本发明能够在音频信号的稀疏并行编码中结合感知听觉掩蔽。
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公开(公告)号：US20120316886A1
公开(公告)日：2012-12-13
申请号：US13491776
申请日：2012-06-08
申请人： Ramin Pishehvar , Hossein Najaf-Zadeh , Frederic Mustiere , Christopher Srinivasa , Hassan Lahdili , Louis Thibault
发明人： Ramin Pishehvar , Hossein Najaf-Zadeh , Frederic Mustiere , Christopher Srinivasa , Hassan Lahdili , Louis Thibault
IPC分类号： G10L19/00
CPC分类号： G10L19/032
摘要： The invention relates to a method and apparatus for efficient encoding of media signals including audio. A 2d sparse representation, or spikegram, of one frame of a digitized audio signal is generated using an overcomplete set of kernels. The spikegram is then mapped to a non-negative matrix, which is decomposed into a 3D component matrix containing hidden components and a 3D weight matrix using a two-dimensional non-negative matrix factorization. Elements of the 3D component and weight matrices are then adaptively quantized using integer programming to determine an optimal quantization scheme, and the quantized values are the optionally encoded using an arithmetic coder.
摘要翻译： 本发明涉及一种用于对包括音频的媒体信号进行有效编码的方法和装置。 使用不完整的内核集合生成数字化音频信号的一帧的2d稀疏表示或尖峰图。 然后将尖峰映射到非负矩阵，其被分解成包含隐藏分量的3D分量矩阵和使用二维非负矩阵因式分解的3D权重矩阵。 然后使用整数规划来自适应地量化3D分量和权重矩阵的元素以确定最佳量化方案，并且量化值可选地使用算术编码器进行编码。
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公开(公告)号：US20080219466A1
公开(公告)日：2008-09-11
申请号：US12073660
申请日：2008-03-07
IPC分类号： H04R3/02
CPC分类号： G10L19/0212 , G10L19/032
摘要： A biologically-inspired process for universal audio coding based on neural spikes is presented. The process is based on the generation of sparse two-dimensional time-frequency representations of audio signals, called spikegrams. The spikegrams are generated by projecting the audio signal onto a set of over-complete adaptive gamma-chirp kernels. A masking model is applied to the spikegrams to remove inaudible spikes and to increase the coding efficiency. In respect of one aspect of the invention, the masked spikegram is then quantized using a genetic-algorithm-based quantizer (or its simplified linear version). The values are then differentially coded using graph based optimization and entropy coded afterwards.
摘要翻译： 提出了一种基于神经尖峰的通用音频编码的生物学启发过程。 该过程基于产生称为尖峰图像的音频信号的稀疏二维时间 - 频率表示。 通过将音频信号投影到一组过完整的自适应伽玛啁啾内核上来产生尖峰图。 将掩蔽模型应用于尖峰图以去除不可听见的尖峰并提高编码效率。 关于本发明的一个方面，然后使用基于遗传算法的量化器（或其简化的线性版本）量化掩蔽的尖峰图。 然后使用基于图的优化和随后的熵编码来对这些值进行差分编码。
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公开(公告)号：US09399475B2
公开(公告)日：2016-07-26
申请号：US14361506
申请日：2011-11-30
摘要： A traction control assembly for connection between a monorail bogie frame and a monorail car. The traction control assembly comprising a first traction link pivotally connected to a first bell crank mechanism and a second traction link pivotally connected to a second bell crank mechanism. The first traction link and the second traction link are capable of absorbing traction forces applied to the monorail bogie. The traction control assembly further comprises a cross link interconnecting the first bell crank mechanism and the second bell crank mechanism and a passive steering assist device interconnecting the first bell crank mechanism and the second bell crank mechanism. The steering assist device causes the traction control assembly to insert shear forces on the monorail bogie during travel of the monorail bogie over a curved section of monorail track for facilitating rotational motion between the monorail bogie and the monorail car.
摘要翻译： 用于单轨转向架和单轨车之间连接的牵引力控制组件。 牵引控制组件包括枢转地连接到第一钟形曲柄机构的第一牵引连杆和枢转地连接到第二曲柄机构的第二牵引连杆。 第一牵引连杆和第二牵引连杆能够吸收施加到单轨转向架的牵引力。 牵引控制组件还包括互连第一钟形曲柄机构和第二钟形曲柄机构的交叉连杆以及将第一钟形曲柄机构和第二曲柄机构相互连接的主动转向辅助装置。 转向辅助装置使得牵引力控制组件在单轨转向架行驶在单轨轨道的弯曲部分的行驶期间将剪力插入到单轨转向架上，以便于单轨转向架和单轨车之间的旋转运动。
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公开(公告)号：US07995688B2
公开(公告)日：2011-08-09
申请号：US12073687
申请日：2008-03-07
申请人： Zhihong Hong , Liang Zhang , Louis Thibault
发明人： Zhihong Hong , Liang Zhang , Louis Thibault
CPC分类号： H04L25/03821 , H04L25/022 , H04L25/0236 , H04L25/025 , H04L25/03324 , H04L25/03828 , H04L27/2647 , H04L2025/03414 , H04L2025/0349
摘要： An iterative channel estimation and inter-carrier interference (ICI) cancellation process is provided for OFDM receivers, and more particularly for mobile OFDM receivers. The iterative process uses decision feedback to estimate both the channel gain and the ICI gains, the latter being the multiplicative gain applied to the adjacent sub-carriers. Thus the receiver performs equalization and ICI cancellation in an iterative fashion and is advantageous for estimating fast fading channels.
摘要翻译： 为OFDM接收机提供迭代信道估计和载波间干扰（ICI）消除过程，更具体地针对移动OFDM接收机。 迭代过程使用判决反馈来估计信道增益和ICI增益，后者是应用于相邻子载波的乘法增益。 因此，接收机以迭代方式执行均衡和ICI消除，并且对于估计快速衰落信道是有利的。
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