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51. 发明申请

WO2017220528A1 AUDIO DECODER AND METHOD FOR TRANSFORMING A DIGITAL AUDIO SIGNAL FROM A FIRST TO A SECOND FREQUENCY DOMAIN 审中-公开
标题翻译：音频解码器和将数字音频信号从第一频率域变换到第二频率域的方法
公开(公告)号：WO2017220528A1
公开(公告)日：2017-12-28
申请号：PCT/EP2017/065011
申请日：2017-06-20
申请人： DOLBY INTERNATIONAL AB
发明人： EKSTRAND, Per , THESING, Robin , VILLEMOES, Lars
IPC分类号： G10L21/00 , G10L21/0388
摘要： There is provided an audio decoder and a method therein for transforming a digital audio signal from a first frequency domain to a second frequency domain. For each received frame of the digital audio signal, the method identifies an upper limit of the frequency range, and if the upper limit of the frequency range is below the Nyquist frequency of said frame of the digital audio signal by more than a threshold amount, the Nyquist frequency of said frame of the digital audio signal is lowered from its original value to a reduced value by removing spectral bands of said frame of the digital audio signal above the identified upper limit of the frequency range. Thereafter said frame of the digital audio signal is transformed from the first frequency domain to the second frequency domain via an intermediate time domain.
摘要翻译：提供了一种音频解码器及其中用于将数字音频信号从第一频域变换到第二频域的方法。对于数字音频信号的每个接收帧，该方法识别频率范围的上限，并且如果频率范围的上限低于数字音频信号的所述帧的奈奎斯特频率大于阈值量，通过去除数字音频信号的所述帧的频谱带高于所识别的频率范围的上限，数字音频信号的所述帧的奈奎斯特频率从其原始值降低到减小的值。此后，数字音频信号的所述帧经由中间时域从第一频域变换到第二频域。

52. 发明申请

WO2016149015A1 DECODING AUDIO BITSTREAMS WITH ENHANCED SPECTRAL BAND REPLICATION METADATA IN AT LEAST ONE FILL ELEMENT 审中-公开
标题翻译：用最小的一个填充元素解码具有增强的光谱带复制音频元素
公开(公告)号：WO2016149015A1
公开(公告)日：2016-09-22
申请号：PCT/US2016/021666
申请日：2016-03-10
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： VILLEMOES, Lars , PURNHAGEN, Heiko , EKSTRAND, Per
IPC分类号： G10L19/00
CPC分类号： G10L19/167 , G10L19/035 , G10L19/24 , G10L21/038
摘要： Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.
摘要翻译：实施例涉及一种包括缓冲器，比特流有效载荷去格局化器和解码子系统的音频处理单元。缓冲器存储编码音频比特流的至少一个块。该块包括一个填充元素，以填充数据的标识符开始。填充数据包括至少一个标志，标识是否对块的音频内容执行增强的频谱带复制（eSBR）处理。还提供了用于解码编码音频比特流的相应方法。

53. 发明申请

WO2015059153A1 PARAMETRIC RECONSTRUCTION OF AUDIO SIGNALS 审中-公开
标题翻译：音频信号的参数重构
公开(公告)号：WO2015059153A1
公开(公告)日：2015-04-30
申请号：PCT/EP2014/072570
申请日：2014-10-21
申请人： DOLBY INTERNATIONAL AB
发明人： VILLEMOES, Lars , LEHTONEN, Heidi-Maria , PURNHAGEN, Heiko , HIRVONEN, Toni
IPC分类号： G10L19/008
CPC分类号： G10L19/167 , G10L19/008 , G10L19/265 , H04S5/005 , H04S2420/03
摘要： An encoding system (400) encodes an N -channel audio signal ( X ), wherein N ≥ 3, as a single-channel downmix signal ( Y ) together with dry and wet upmix parameters (C, P ). In a decoding system (200), a decorrelating section (101 ) outputs, based on the downmix signal, an ( N -1)-channel decorrelated signal ( Z ); a dry upmix section (102) maps the downmix signal linearly in accordance with dry upmix coefficients ( C ) determined based on the dry upmix parameters; a wet upmix section (103) populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients ( P ) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section (104) combines outputs from the upmix sections to obtain a reconstructed signal ( X ) corresponding to the signal to be reconstructed.
摘要翻译：编码系统（400）将干扰和湿混合参数（C，P）的N沟道音频信号（X）编码为N≥3，作为单通道下混信号（Y）。在解码系统（200）中，解相关部分（101）根据所述下混信号输出（N-1）信道解相关信号（Z）; 干混合部分（102）根据干燥上混参数确定的干混上限系数（C）线性地映射下混合信号; 湿混合部分（103）基于湿混合参数填充中间矩阵，并且知道中间矩阵属于预定矩阵类，通过将中间矩阵乘以预定义矩阵来获得湿上混系数（P），并将去相关信号根据潮湿混合系数线性地进行; 并且组合部分（104）组合来自上混段的输出，以获得与要重构的信号相对应的重构信号（X）。

54. 发明申请

WO2015059152A1 DECORRELATOR STRUCTURE FOR PARAMETRIC RECONSTRUCTION OF AUDIO SIGNALS 审中-公开
标题翻译：参数重构音频信号的装饰结构
公开(公告)号：WO2015059152A1
公开(公告)日：2015-04-30
申请号：PCT/EP2014/072568
申请日：2014-10-21
申请人： DOLBY INTERNATIONAL AB
发明人： VILLEMOES, Lars , HIRVONEN, Toni , PURNHAGEN, Heiko
IPC分类号： G10L19/008
CPC分类号： H04S7/30 , G10L19/002 , G10L19/008 , G10L25/21 , H04S2400/03 , H04S2420/03
摘要： An encoding system encodes multiple audio signals (X) as a downmix signal (Y) together with wet and dry upmix coefficients (P, C). In a decoding system, a pre-multiplier (101) computes an intermediate signal (W) by mapping the downmix signal linearly in accordance with a first set of coefficients (Q); a decorrelating section (102) outputs a decorrelated signal (Z) based on the intermediate signal; a wet upmix section (103) computes a wet upmix signal by mapping the decorrelated signal linearly in accordance with the wet upmix coefficients; a dry upmix section (104) computes a dry upmix signal by mapping the downmix signal linearly in accordance with the dry upmix coefficients; a combining section (105) provides a multidimensional reconstructed signal (X) by combining the wet and dry upmix signals; and a converter (106) computes the first set of coefficients based on the wet and dry upmix coefficients and supplies this to the pre-multiplier.
摘要翻译：编码系统将多个音频信号（X）与湿和干混合系数（P，C）一起编码为下混合信号（Y）。在解码系统中，预乘法器（101）通过根据第一组系数（Q）线性地映射下混合信号来计算中间信号（W）; 去相关部分（102）基于中间信号输出解相关信号（Z）; 湿混合部分（103）通过根据湿混合系数线性地映射去相关信号来计算湿上混信号; 干混合部分（104）通过根据干燥上混系数线性地映射下混合信号来计算干混合信号; 组合部分（105）通过组合湿和干混合信号来提供多维重建信号（X）; 并且转换器（106）基于湿和干混合系数来计算第一组系数，并将其提供给预乘数。

55. 发明申请

WO2014187987A1 METHODS FOR AUDIO ENCODING AND DECODING, CORRESPONDING COMPUTER-READABLE MEDIA AND CORRESPONDING AUDIO ENCODER AND DECODER 审中-公开
标题翻译：音频编码和解码方法，相关的可编程媒体和相关的音频编码器和解码器
公开(公告)号：WO2014187987A1
公开(公告)日：2014-11-27
申请号：PCT/EP2014/060728
申请日：2014-05-23
申请人： DOLBY INTERNATIONAL AB
发明人： PURNHAGEN, Heiko , VILLEMOES, Lars , SAMUELSSON, Leif Jonas , HIRVONEN, Toni
IPC分类号： G10L19/008 , G10L19/20 , H04S7/00
CPC分类号： G10L19/008 , G10L19/20 , H04S3/02 , H04S5/00 , H04S7/30 , H04S2400/03 , H04S2400/11 , H04S2420/03 , H04S2420/07
摘要： The present disclosure provides methods, devices and computer program products which provide less complex and more flexible control of the introduced decorrelation in an audio coding system. According to the disclosure, this is achieved by calculating and using two weighting factors, one for an approximated audio object and one for a decorrelated audio object, for introduction of decorrelation of audio objects in the audio coding system.
摘要翻译：本公开提供了在音频编码系统中提供对引入的去相关的较不复杂和更灵活的控制的方法，设备和计算机程序产品。根据本公开，这是通过计算和使用两个加权因子来实现的，其中一个用于近似音频对象，一个用于解相关音频对象，用于引入音频编码系统中音频对象的去相关。

56. 发明申请

WO2013186344A2 SMOOTH CONFIGURATION SWITCHING FOR MULTICHANNEL AUDIO RENDERING BASED ON A VARIABLE NUMBER OF RECEIVED CHANNELS 审中-公开
标题翻译：基于可变数量的接收频道进行多通道音频渲染的最佳配置切换
公开(公告)号：WO2013186344A2
公开(公告)日：2013-12-19
申请号：PCT/EP2013/062340
申请日：2013-06-14
申请人： DOLBY INTERNATIONAL AB
发明人： PURNHAGEN, Heiko , SEHLSTROM, Leif , ROEDEN, Karl, Jonas , KJOERLING, Kristofer , VILLEMOES, Lars
IPC分类号： G10L19/18
CPC分类号： G10L19/008 , G10L19/0017 , G10L19/18 , H04S3/008 , H04S2400/03 , H04S2420/03
摘要： A decoding system reconstructs an n-channel audio signal on the basis of an input signal representing the audio signal, in different time frames, either by parametric coding or as n discretely coded channels. Parametric decoding uses a core signal and mixing parameters controlling a spatial synthesis stage, to which a downmix signal is supplied from a downmix stage. The downmix stage realizes a projection on the downmix signal based on an n- channel input signal, either a discretely coded signal or a core signal padded with neutral-valued channels. The padding may take place either on the decoding side (reduced parametric coding) or the encoding side. In an embodiment, an audio decoder (110) in the decoding system pads the core signal during an initial portion of each reduced parametrically coded time frame directly succeeding a discretely coded time frame and during a final portion of each reduced parametrically coded time frame directly preceding a discretely coded time frame.
摘要翻译：解码系统通过参数编码或作为n个离散编码的信道，在不同的时间帧中，基于表示音频信号的输入信号来重构n信道音频信号。参数解码使用核心信号和控制空间合成阶段的混合参数，下混合信号从下混合级提供给它。下混合阶段基于n-沟道输入信号，即离散编码信号或填充有中性信道的核心信号实现对下混信号的投影。填充可以在解码侧（缩减参数编码）或编码侧进行。在一个实施例中，解码系统中的音频解码器（110）在直接在离散编码的时间帧之后的每个缩减的参数编码时间帧的初始部分期间以及在直接前进的每个缩减的参数编码时间帧的最后部分期间，离散编码的时间帧。

57. 发明申请

WO2013142650A1 ENABLING SAMPLING RATE DIVERSITY IN A VOICE COMMUNICATION SYSTEM 审中-公开
标题翻译：在语音通信系统中实现采样速率多样性
公开(公告)号：WO2013142650A1
公开(公告)日：2013-09-26
申请号：PCT/US2013/033228
申请日：2013-03-21
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING CORPORATION
发明人： PURNHAGEN, Heiko , SEHLSTROM, Leif , VILLEMOES, Lars , DICKINS, Glenn N. , VINTON, Mark S.
IPC分类号： G10L19/16 , H04M3/56
CPC分类号： G10L19/03 , G10L19/002 , G10L19/16 , G10L19/26 , H04M3/56 , H04M3/568
摘要： An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.
摘要翻译：音频通信端点接收包含表示音频信号的频谱内容的频谱分量的比特流，其中频谱分量涉及延伸到第一中断频率的第一范围，高于该频率分量的任何频谱分量未被分配。端点通过去除频谱分量或增加与第一和第二断开频率之间的范围有关的中性频谱分量，根据延伸到第二中断频率的第二范围来适配所接收的比特流。然后，端点衰减第一和第二断裂频率中最小的邻域中的频谱内容，从而实现逐渐的频谱衰减。之后，通过在适配和衰减的接收比特流中与所述第二范围有关的频谱分量上的逆变换来重构音频信号。在较小的计算费用下，端点可以适应接收到的比特流中的不同采样率。

58. 发明申请

WO2013124443A1 LOW DELAY REAL-TO-COMPLEX CONVERSION IN OVERLAPPING FILTER BANKS FOR PARTIALLY COMPLEX PROCESSING 审中-公开
标题翻译：用于部分复杂加工的过渡过滤器中的低延迟实时复合转换
公开(公告)号：WO2013124443A1
公开(公告)日：2013-08-29
申请号：PCT/EP2013/053607
申请日：2013-02-22
申请人： DOLBY INTERNATIONAL AB
发明人： VILLEMOES, Lars , MUNDT, Harald
IPC分类号： H03H17/02
CPC分类号： H03G3/00 , H03H17/0266
摘要： An arrangement of overlapping filter banks comprises a synthesis stage and an analysis stage. The synthesis stage receives a first signal segmented into time blocks and outputs, based thereon, an intermediate signal to be received by the analysis stage forming the basis for the computation of a second signal segmented into time frames. In an embodiment, the synthesis stage is operable to release an approximate value of the intermediate signal in a time block located L - 1 time blocks ahead of its output block, which approximate value is computed on the basis of any available time blocks of the first signal, so that the approximate value contributes, in the analysis stage, to the second signal. The delay is typically reduced by L - 1 blocks. Applications include audio signal processing in general and real -to-complex conversion in particular.
摘要翻译：重叠滤波器组的布置包括合成阶段和分析阶段。合成级接收分段成时间块的第一信号，并输出由分析级接收的中间信号，该中间信号形成用于计算分段成时间帧的第二信号的基础。在一个实施例中，合成阶段可操作以在位于其输出块之前的L-1个时间块的时间块中释放中间信号的近似值，该近似值是基于第一个信号，使得近似值在分析阶段中有助于第二信号。延迟通常减少L-1个块。应用包括一般的音频信号处理，特别是真实到复杂的转换。

59. 发明申请

WO2022078960A1 SIGNAL CODING USING A GENERATIVE MODEL AND LATENT DOMAIN QUANTIZATION 审中-公开
公开(公告)号：WO2022078960A1
公开(公告)日：2022-04-21
申请号：PCT/EP2021/078053
申请日：2021-10-11
申请人： DOLBY INTERNATIONAL AB
发明人： KLEJSA, Janusz , VILLEMOES, Lars , HEDELIN, Per
IPC分类号： G10L19/02 , G06N3/04 , G06N3/0454
摘要： The present disclosure provides a decoder configured to receive a finite bitrate stream that includes a quantized latent frame, where the quantized latent frame includes a quantized representation of a current frame of a signal in a latent domain different from a first domain; to generate a reconstructed latent frame from the quantized latent frame; to use a generative neural network model to perform a task for which the general neural network model has been trained, wherein the task includes to generate parameters for an invertible mapping from the latent domain to the first domain; to reconstruct a current frame of the signal in the first domain, which includes to map the reconstructed latent frame to the first domain by use of the invertible mapping, and to use the reconstructed current frame of the signal in the first domain to update a state of the generative neural network model.

60. 发明申请

WO2016146492A1 DECODING AUDIO BITSTREAMS WITH ENHANCED SPECTRAL BAND REPLICATION METADATA IN AT LEAST ONE FILL ELEMENT 审中-公开
标题翻译：在至少一个填充元素中使用增强的谱带复制元数据解码音频比特流
公开(公告)号：WO2016146492A1
公开(公告)日：2016-09-22
申请号：PCT/EP2016/055202
申请日：2016-03-10
申请人： DOLBY INTERNATIONAL AB
发明人： VILLEMOES, Lars , PURNHAGEN, Heiko , EKSTRAND, Per
IPC分类号： G10L19/16 , G10L19/24 , G10L21/038 , G10L19/035
CPC分类号： G10L19/167 , G10L19/035 , G10L19/24 , G10L21/038
摘要： Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.
摘要翻译：实施例涉及包括缓冲器，比特流有效负载解格式化器和解码子系统的音频处理单元。该缓冲器存储经编码的音频比特流的至少一个块。该块包含一个填充元素，该元素以一个标识符开头，后面跟着填充数据。填充数据包括至少一个标识标识是否对块的音频内容执行增强的频谱带复制（eSBR）处理。还提供了用于对编码的音频比特流进行解码的相应方法。

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