公开(公告)号：US20120076307A1

公开(公告)日：2012-03-29

申请号：US13375035

申请日：2010-05-31

申请人： Albertus Cornelis Den Brinker ,  Aki Sakari Harma

发明人： Albertus Cornelis Den Brinker ,  Aki Sakari Harma

IPC分类号： H04R5/00

CPC分类号： H04S3/002 ,  G10L19/008

摘要： An audio apparatus comprises a processor (101) for providing a set of audio channels. A prediction circuit (103) generates a predicted signal for a first channel by adaptive filtering of a second channel by an adaptive filter. An adaptation processor (105) adapts the adaptive filter to minimize a cost function indicative of a difference between the predicted signal and the first channel. A compensation processor (107) then generates a non-predicted signal by compensating the first signal for the predicted signal and a distribution processor (109) generates an output set of audio channels by distributing at least the predicted signal and the non-predicted signal over the output set of audio signals where the distribution is different for the predicted signal and the non-predicted signal. The cross-channel predictive filtering provides signal components that represent different spatial characteristics of the originating sound and which are therefore advantageously distributed differently for the output channels.

摘要翻译： 音频装置包括用于提供一组音频通道的处理器（101）。 预测电路（103）通过自适应滤波器对第二信道进行自适应滤波，生成第一信道的预测信号。 适配处理器（105）使自适应滤波器适配，以最小化指示预测信号和第一信道之间的差异的成本函数。 补偿处理器（107）然后通过补偿预测信号的第一信号来产生非预测信号，并且分配处理器（109）通过至少分配预测信号和非预测信号来生成音频信道的输出集合 其中对于预测信号和非预测信号的分布不同的音频信号的输出组。 交叉通道预测滤波提供了表示起始声音的不同空间特征的信号分量，因此有利地为输出声道分配不同的信号分量。

公开(公告)号：US07640156B2

公开(公告)日：2009-12-29

申请号：US10564656

申请日：2004-07-08

申请人： Andreas Johannes Gerrits ,  Albertus Cornelis Den Brinker

发明人： Andreas Johannes Gerrits ,  Albertus Cornelis Den Brinker

IPC分类号： G10L19/00

CPC分类号： G10L19/093

摘要： In a sinusoidal audio encoder a number of sinusoids are estimated per audio segment. A sinusoid is represented y frequency, amplitude and phase. Normally, phase is quantised independent of frequency The invention uses a frequency dependent quantisation of phase, and in particular the low frequencies are quantised using smaller quantisation intervals than at higher frequencies. Thus, the unwrapped phases of the lower frequencies are quantised more accurately, possibly with a smaller quantisation range, than the phases of the higher frequencies. The invention gives a significant improvement in decoded signal quality, especially for low bit-rate quantisers.

摘要翻译： 在正弦音频编码器中，每个音频段估计出多个正弦波。 正弦曲线表示y频率，振幅和相位。 通常，相位与频率无关地量化本发明使用相位的频率相关量化，特别是使用比较高频率更小的量化间隔量化低频。 因此，与较高频率的相位相比，更准确地量化更低频率的展开相位，可能具有较小的量化范围。 本发明给出了解码信号质量的显着改进，特别是对于低比特率量化器。

公开(公告)号：US07587313B2

公开(公告)日：2009-09-08

申请号：US10598796

申请日：2005-03-08

申请人： Andreas Johannes Gerrits ,  Albertus Cornelis Den Brinker

发明人： Andreas Johannes Gerrits ,  Albertus Cornelis Den Brinker

IPC分类号： G10L19/14

CPC分类号： G10L19/20 ,  G10L19/022 ,  G10L19/093

摘要： The method creates an audio stream comprising tracks of sinusoidal components linked across a plurality of sequential time segments. Segments in each track are weighted with a normal window (WI, W2, W3), and consecutive segments have a normal period of overlap (0) of their trailing edges and leading edges. Segments in which a transient5 component is determined are weighted with a first modified window (WIm) having a modified trailing edge, and the following segment in the track is weighted with a second modified window (W2m) having a modified leading edge, so that the modified trailing edge and the modified leading edge have a modified period of overlap (0m) that comprises the transient component and that is shorter than the normal period of overlap (0), and wherein the audio stream includes sinusoidal codes representing the frequency and the transient. According to the invention, the modified period of overlap (0m) depends on the frequency value (f).

摘要翻译： 该方法创建包括跨多个连续时间段链接的正弦分量的轨道的音频流。 每个轨道中的段用正常窗口（WI，W2，W3）加权，并且连续的段具有其后沿和前沿的正常重叠周期（0）。 确定了瞬态5分量的分段用具有修改的后沿的第一修改窗口（WIm）加权，并且轨道中的下一分段被加权具有修改的前沿的第二修改窗口（W2m），使得 经修改的后沿和修改的前沿具有修改的重叠周期（0m），其包括瞬态分量并且短于正常重叠周期（0），并且其中音频流包括表示频率和瞬态的正弦代码 。 根据本发明，修改的重叠周期（0m）取决于频率值（f）。

公开(公告)号：US20090106030A1

公开(公告)日：2009-04-23

申请号：US11718613

申请日：2005-11-02

申请人： Albertus Cornelis Den Brinker ,  Felipe Riera Palou

发明人： Albertus Cornelis Den Brinker ,  Felipe Riera Palou

IPC分类号： G10L19/00

CPC分类号： G10L19/06 ,  G10L19/093

摘要： There is described a method of encoding a signal (s(n)) in a coder (400) to generate a corresponding encoded bit-stream (x(n); STP). The method comprises steps of: (a) processing the signal (s(n)) to determine main sinusoidal components and transient components thereof to generate corresponding component parameters; (b) processing the signal (s(n)) by removing the sinusoidal and transient components therefrom to generate a residual signal (r(n)); (c) processing the residual signal (r(n)) to determine a spectral representation (PSD) and determining therefrom a spectral broadening measure (SBM); (d) determining from the residual signal (r(n)) spectral envelope parameters by linear prediction; and (e) combining the components parameters together with the spectral envelope parameters and the spectral broadening measure to generate the encoded bit-stream. The method is capable of reducing noise that would otherwise arise were the bitstream to be decoded not subjected to such spectral broadening.

摘要翻译： 描述了在编码器（400）中编码信号（s（n））以产生相应的编码比特流（x（n）; STP）的方法。 该方法包括以下步骤：（a）处理信号（s（n））以确定主正弦分量及其瞬态分量以产生相应的分量参数; （b）通过从其中去除正弦和瞬态分量来处理信号（s（n））以产生残差信号（r（n））; （c）处理残余信号（r（n））以确定频谱表示（PSD）并由此确定频谱展宽测量（SBM）; （d）通过线性预测从剩余信号（r（n））频谱包络参数确定; 和（e）将分量参数与频谱包络参数和频谱展宽测量结合在一起以产生编码比特流。 该方法能够减少否则将要解码的比特流不经受这种频谱展宽的噪声。

公开(公告)号：US20080215330A1

公开(公告)日：2008-09-04

申请号：US11996364

申请日：2006-07-18

申请人： Aki Sakari Harma ,  Albertus Cornelis Den Brinker

发明人： Aki Sakari Harma ,  Albertus Cornelis Den Brinker

IPC分类号： G10L13/00

CPC分类号： G10L21/00 ,  G10L19/06 ,  G10L2021/0135 ,  H04R25/353

摘要： A method of modifying an audio signal comprises the steps of analyzing the input audio signal (x) so as to produce a set of filter parameters (p) and a residual signal (r), modifying the set of filter parameters (p) so as to produce a modified set of filter parameters (p′), and synthesizing an output audio signal (y) using the modified set of filter parameters (p′) and the residual signal (r). The set of filter parameters (p) comprises poles (λA) and coefficients (a; c). The step of modifying the filter parameters (p) involves interpolating lattice filter reflection coefficients (c) so as to scale the spectral envelope of the audio signal.

摘要翻译： 修改音频信号的方法包括分析输入音频信号（x）以产生一组滤波器参数（p）和残差信号（r）的步骤，修改滤波器参数集合（p），以便 以产生经修改的滤波器参数集合（p'），并且使用修改的滤波器参数（p'）和残余信号（r）来合成输出音频信号（y）。 滤波器参数（p）的集合包括极（λΛA）和系数（a; c）。 修改滤波器参数（p）的步骤涉及内插滤波器反射系数（c），以便缩放音频信号的频谱包络。

公开(公告)号：US20080184871A1

公开(公告)日：2008-08-07

申请号：US11908321

申请日：2006-02-01

申请人： Marek Szczerba ,  Albertus Cornelis Den Brinker ,  Andreas Johannes Gerrits ,  Arnoldus Werner Johannes Oomen ,  Marc Klein Middelink

发明人： Marek Szczerba ,  Albertus Cornelis Den Brinker ,  Andreas Johannes Gerrits ,  Arnoldus Werner Johannes Oomen ,  Marc Klein Middelink

IPC分类号： G10H1/06

CPC分类号： G10H7/00 ,  G10H1/22 ,  G10H2230/041 ,  G10H2250/495

摘要： A device (1) is arranged for synthesizing sound represented by sets of parameters, each set comprising noise parameters (NP) representing noise components of the sound and optionally also other parameters representing other components, such as transients and sinusoids. Each set of parameters may correspond with a sound channel, such as a MIDI voice. In order to reduce the computational load, the device comprises a selection unit (2) for selecting a limited number of sets from the total number of sets on the basis of a perceptual relevance value, such as the amplitude or energy. The device further comprises a synthesizing unit (3) for synthesizing the noise components using the noise parameters of the selected sets only.

摘要翻译： 设备（1）被布置为用于合成由参数集合表示的声音，每组包括表示声音的噪声分量的噪声参数（NP），以及可选地还包括表示其他分量的其它参数，例如瞬变和正弦波。 每组参数可以对应于诸如MIDI声音之类的声音通道。 为了减少计算负荷，该装置包括一个选择单元（2），用于根据诸如振幅或能量之类的感知相关性值从总数量集中选择有限数量的集合。 该装置还包括用于仅使用所选集合的噪声参数合成噪声分量的合成单元（3）。

公开(公告)号：US06925434B2

公开(公告)日：2005-08-02

申请号：US09804022

申请日：2001-03-12

申请人： Arnoldus Werner Johannes Oomen ,  Albertus Cornelis Den Brinker

发明人： Arnoldus Werner Johannes Oomen ,  Albertus Cornelis Den Brinker

IPC分类号： G10L19/02 ,  G10L21/06

CPC分类号： G10L19/02

摘要： Coding (1) of an audio signal is provided including estimating (110) a position of a transient signal component in the audio signal, matching (111,112) a shape function on the transient signal component in case the transient signal component is gradually declining after an initial increase, which shape function has a substantially exponential initial behavior and a substantially logarithmic declining behavior; and including (15) the position and shape parameters describing the shape function in an audio stream (AS).

公开(公告)号：US09070358B2

公开(公告)日：2015-06-30

申请号：US13505758

申请日：2010-11-05

申请人： Albertus Cornelis Den Brinker ,  Erik Gosuinus Petrus Schuijers ,  Arnoldus Werner Johanees Oomen

发明人： Albertus Cornelis Den Brinker ,  Erik Gosuinus Petrus Schuijers ,  Arnoldus Werner Johanees Oomen

IPC分类号： H04R5/00 ,  G10L19/008

CPC分类号： G10L19/008

摘要： An encoder for a multi-channel audio signal which comprises a down-mixer (201, 203, 205) for generating a down-mix as a combination of at least a first and second channel signal weighted by respectively a first and second weight with different amplitudes for at least some time-frequency intervals. Furthermore, a circuit (201, 203, 209) generates up-mix parametric data characterizing a relationship between the channel signals as well as characterizing the weights. A circuit generates weight estimates for the encoder weights from the up-mix parametric data; and comprises an up-mixer (407) which recreates the multi channel audio signal by up-mixing the down-mix in response to the up-mix parametric data, the first weight estimate and the second weight estimate. The up-mixing is dependent on the amplitude of at least one of the weight estimate(s).

摘要翻译： 一种用于多声道音频信号的编码器，包括用于产生下混合器的下混频器（201,203,205），所述下混频器作为至少第一和第二声道信号的组合，分别由不同的第一和第二加权器加权 至少一些时间间隔的振幅。 此外，电路（201,203,209）产生表征信道信号之间的关系以及表征权重的混合参数数据。 电路从混合参数数据生成编码器权重的权重估计; 并且包括上混频器（407），其通过响应于混合参数数据，第一权重估计和第二权重估计，通过向下混合来重新创建多声道音频信号。 上混合取决于至少一个重量估计的振幅。

公开(公告)号：US08346564B2

公开(公告)日：2013-01-01

申请号：US11909730

申请日：2006-03-16

申请人： Gerard Herman Hotho ,  Dirk Jeroen Breebaart ,  Erik Gosuinus Petrus Schuijers ,  Albertus Cornelis Den Brinker ,  Lars Falck Villemoes ,  Heiko Purnhagen ,  Karl Jonas Roden

发明人： Gerard Herman Hotho ,  Dirk Jeroen Breebaart ,  Erik Gosuinus Petrus Schuijers ,  Albertus Cornelis Den Brinker ,  Lars Falck Villemoes ,  Heiko Purnhagen ,  Karl Jonas Roden

IPC分类号： G10L19/00 ,  G10L21/04

CPC分类号： G10L19/008 ,  H04R2420/07 ,  H04S3/008

摘要： A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi-channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down-mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102). In the decoder (20), first the spatial down-mix (202) is reconstructed from the alternative down-mix (103) and the parameters (105). Next, the spatial down-mix (202) is decoded into the multi-channel audio signal (203).

摘要翻译： 一种用于对多声道音频信号（101）进行编码的多声道音频编码器（10）。 5.1声道音频信号转换成空间缩混（102）。 立体声信号和相关参数（104,105）。 编码器（10）包括第一和第二单元（110,120）。 第一单元（110）将多声道音频信号（101）编码成空间缩混（102）和参数（104）。 这些参数（104）使得多声道解码器（20）能够从空间缩混（102）重建多声道音频信号（203）。 第二单元（120）从空间缩混（102）生成参数（105），使参数（105）能够使解码器从替代的下混合（103）重构空间缩混（202）。 所谓的艺术混合，已经在一个健全的工作室手工混合。 以这种方式，解码器（20）可以有效地处理接收替代的下混合（103）而不是常规的空间，混合（102）的情况。 在解码器（20）中，首先，从替代的下混合（103）和参数（105）重构空间缩混（202）。 接下来，空间缩混（202）被解码为多声道音频信号（203）。

公开(公告)号：US20120224702A1

公开(公告)日：2012-09-06

申请号：US13505758

申请日：2010-11-05

申请人： Albertus Cornelis Den Brinker ,  Erik Gosuinus Petrus Schuijers ,  Arnoldus Werner Johanees Oomen

发明人： Albertus Cornelis Den Brinker ,  Erik Gosuinus Petrus Schuijers ,  Arnoldus Werner Johanees Oomen

IPC分类号： H04R5/00

CPC分类号： G10L19/008

摘要： An encoder for a multi-channel audio signal which comprises a down-mixer (201, 203, 205) for generating a down-mix as a combination of at least a first and second channel signal weighted by respectively a first and second weight with different amplitudes for at least some time-frequency intervals. Furthermore, a circuit (201, 203, 209) generates up-mix parametric data characterizing a relationship between the channel signals as well as characterizing the weights. A circuit generates weight estimates for the encoder weights from the up-mix parametric data; and comprises an up-mixer (407) which recreates the multi channel audio signal by up-mixing the down-mix in response to the up-mix parametric data, the first weight estimate and the second weight estimate. The up-mixing is dependent on the amplitude of at least one of the weight estimate(s).

摘要翻译： 一种用于多声道音频信号的编码器，包括用于产生下混合器的下混频器（201,203,205），所述下混频器作为至少第一和第二声道信号的组合，分别由具有不同 至少一些时间间隔的振幅。 此外，电路（201,203,209）产生表征信道信号之间的关系以及表征权重的混合参数数据。 电路从混合参数数据生成编码器权重的权重估计; 并且包括上混频器（407），其通过响应于混合参数数据，第一权重估计和第二权重估计，通过向下混合来重新创建多声道音频信号。 上混合取决于至少一个重量估计的幅度。