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1. 发明申请

WO2021252823A1 METHODS, APPARATUS, AND SYSTEMS FOR DETECTION AND EXTRACTION OF SPATIALLY-IDENTIFIABLE SUBBAND AUDIO SOURCES 审中-公开
公开(公告)号：WO2021252823A1
公开(公告)日：2021-12-16
申请号：PCT/US2021/036900
申请日：2021-06-11
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： MASTER, Aaron Steven , LU, Lie , MUNDT, Harald
IPC分类号： G10L21/0272
摘要： In an embodiment, a method comprises: transforming one or more frames of a two-channel time domain audio signal into a time-frequency domain representation including a plurality of time-frequency tiles, wherein the frequency domain of the time-frequency domain representation includes a plurality of frequency bins grouped into subbands. For each time-frequency tile, the method comprises: calculating spatial parameters and a level for the time-frequency tile; modifying the spatial parameters using shift and squeeze parameters; obtaining a softmask value for each frequency bin using the modified spatial parameters, the level and subband information; and applying the softmask values to the time-frequency tile to generate a modified time-frequency tile of an estimated audio source. In an embodiment, a plurality of frames of the time-frequency tiles are assembled into a plurality of chunks, wherein each chunk includes a plurality of subbands, and the method described above is performed on each subband of each chunk.

2. 发明申请

WO2017087650A1 HEADTRACKING FOR PARAMETRIC BINAURAL OUTPUT SYSTEM AND METHOD 审中-公开
标题翻译：参数化骨门输出系统的头部追踪及方法
公开(公告)号：WO2017087650A1
公开(公告)日：2017-05-26
申请号：PCT/US2016/062497
申请日：2016-11-17
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： BREEBAART, Dirk Jeroen , COOPER, David Matthew , DAVIS, Mark F. , MCGRATH, David S. , KJOERLING, Kristofer , MUNDT, Harald , WILSON, Rhonda J.
IPC分类号： H04S3/00
CPC分类号： H04S3/004 , G10L19/008 , H04S7/304 , H04S2400/01 , H04S2400/11 , H04S2420/03
摘要： A method of encoding channel or object based input audio for playback, the method including the steps of: (a) initially rendering the channel or object based input audio into an initial output presentation; (b) determining an estimate of the dominant audio component from the channel or object based input audio and determining a series of dominant audio component weighting factors for mapping the initial output presentation into the dominant audio component; (c) determining an estimate of the dominant audio component direction or position; and (d) encoding the initial output presentation, the dominant audio component weighting factors, the dominant audio component direction or position as the encoded signal for playback.
摘要翻译：一种对用于回放的基于声道或对象的输入音频进行编码的方法，所述方法包括以下步骤：（a）最初将基于声道或基于对象的输入音频呈现为初始输出呈现; （b）从所述基于声道或基于对象的输入音频确定所述主导音频分量的估计，并确定用于将所述初始输出呈现映射到所述主导音频分量的一系列主导音频分量加权因子; （c）确定主要音频分量方向或位置的估计; （d）将初始输出呈现，主要音频分量加权因子，主要音频分量方向或位置编码为用于回放的编码信号。

3. 发明申请

WO2022120093A1 IMMERSIVE VOICE AND AUDIO SERVICES (IVAS) WITH ADAPTIVE DOWNMIX STRATEGIES 审中-公开
公开(公告)号：WO2022120093A1
公开(公告)日：2022-06-09
申请号：PCT/US2021/061671
申请日：2021-12-02
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： MUNDT, Harald , MCGRATH, David S. , TYAGI, Rishabh
IPC分类号： G10L19/008 , H04S5/00 , G10L19/24
摘要： Disclosed is an audio signal encoding/decoding method that uses an encoding downmix strategy applied at an encoder that is different than a decoding re-mix/upmix strategy applied at a decoder. Based on the type of downmix coding scheme, the method comprises: computing input downmixing gains to be applied to the input audio signal to construct a primary downmix channel; determining downmix scaling gains to scale the primary downmix channel; generating prediction gains based on the input audio signal, the input downmixing gains and the downmix scaling gains; determining residual channel(s) from the side channels by using the primary downmix channel and the prediction gains to generate side channel predictions and subtracting the side channel predictions from the side channels; determining decorrelation gains based on energy in the residual channels; encoding the primary downmix channel, the residual channel(s), the prediction gains and the decorrelation gains; and sending the bitstream to a decoder.

4. 发明申请

WO2014052745A1 NEAR-END INDICATION THAT THE END OF SPEECH IS RECEIVED BY THE FAR END IN AN AUDIO OR VIDEO CONFERENCE 审中-公开
标题翻译：在音频或视频会议末尾接收到语音结束的最终指示
公开(公告)号：WO2014052745A1
公开(公告)日：2014-04-03
申请号：PCT/US2013/062159
申请日：2013-09-27
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： SHI, Dong , SUN, Xuejing , LI, Kai , HUANG, Shen , MUNDT, Harald , PURNHAGEN, Heiko , DICKINS, Glenn
IPC分类号： H04M3/56 , H04M9/08
CPC分类号： H04N7/147 , H04M3/569 , H04M9/082 , H04M2201/12 , H04M2201/14 , H04M2201/38 , H04M2203/258 , H04M2203/352 , H04N7/15
摘要： Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.
摘要翻译：描述用于音频或视频会议的客户端设备和方法的实施例。实施例包括偏移检测单元，配置单元，估计器和输出单元。偏移检测单元检测输入到客户端设备的语音偏移。配置单元确定从客户端设备到每个远端的语音延迟。估计器估计远端用户根据语音延迟感知到偏移的时间。输出单元输出可感知的信号，指示远端的用户基于为远端估计的时间感知偏移量。可感知的信号有助于避免各方之间的冲突。

5. 发明申请

WO2022120011A1 ROTATION OF SOUND COMPONENTS FOR ORIENTATION-DEPENDENT CODING SCHEMES 审中-公开
公开(公告)号：WO2022120011A1
公开(公告)日：2022-06-09
申请号：PCT/US2021/061549
申请日：2021-12-02
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： BRUHN, Stefan , MUNDT, Harald , MCGRATH, David S. , BROWN, Stefanie
IPC分类号： G10L19/008
摘要： Method for encoding scene-based audio is provided. In some implementations, the method involves determining, by an encoder, a spatial direction of a dominant sound component in a frame of an input audio signal. In some implementations, the method involves determining rotation parameters based on the determined spatial direction and a direction preference of a coding scheme to be used to encode the input audio signal. In some implementations, the method involves rotating sound components of the frame based on the rotation parameters such that, after being rotated, the dominant sound component has a spatial direction that aligns with the direction preference of the coding scheme. In some implementations, the method involves encoding the rotated sound components of the frame of the input audio signal using the coding scheme in connection with an indication of the rotation parameters or an indication of the spatial direction of the dominant sound component.

6. 发明申请

WO2013173314A1 EFFICIENT ENCODING AND DECODING OF MULTI-CHANNEL AUDIO SIGNAL WITH MULTIPLE SUBSTREAMS 审中-公开
标题翻译：多通道音频信号的有效编码和解码
公开(公告)号：WO2013173314A1
公开(公告)日：2013-11-21
申请号：PCT/US2013/040919
申请日：2013-05-14
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： MUNDT, Harald , RIEDMILLER, Jeffrey , ROEDEN, Karl J. , WARD, Michael , WILLIAMS, Phillip
IPC分类号： G10L19/008 , G10L19/032 , H04S3/00 , G10L19/24
CPC分类号： G10L19/008 , G10L19/032 , G10L19/24 , H04S3/008
摘要： The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for improving the quality of encoded multi-channel audio signals. An audio encoder configured to encode a multi-channel audio signal according to a total available data-rate is described. The multi-channel audio signal is representable as a basic group (121) of channels for rendering the multi-channel audio signal in accordance to a basic channel configuration, and as an extension group (122) of channels, which - in combination with the basic group (122) - is for rendering the multi-channel audio signal in accordance to an extended channel configuration. The basic channel configuration and the extended channel configuration are different from one another.
摘要翻译：本文件涉及音频编码/解码。特别地，本文件涉及用于提高编码的多声道音频信号的质量的方法和系统。描述了被配置为根据总可用数据速率对多声道音频信号进行编码的音频编码器。多声道音频信号可以被表示为用于根据基本信道配置呈现多声道音频信号的信道的基本组（121），以及作为信道的扩展组（122），其与基本组（122） - 用于根据扩展通道配置呈现多声道音频信号。基本通道配置和扩展通道配置是不同的。

7. 发明申请

WO2019007969A1 LOW COMPLEXITY DENSE TRANSIENT EVENTS DETECTION AND CODING 审中-公开
公开(公告)号：WO2019007969A1
公开(公告)日：2019-01-10
申请号：PCT/EP2018/067970
申请日：2018-07-03
申请人： DOLBY INTERNATIONAL AB
发明人： BISWAS, Arijit , SCHUG, Michael , MUNDT, Harald
IPC分类号： G10L19/025 , G10L19/03
CPC分类号： G10L19/025 , G10L19/03
摘要： The present disclosure relates to methods and apparatus for audio coding. A method of encoding a portion of an audio signal comprises determining whether the portion of the audio signal is likely to contain dense transient events, and if it is determined that the portion of the audio signal is likely to contain dense transient events, quantizing the portion of the audio signal using a quantization 5 mode that applies a substantially constant signal-to-noise ratio over frequency for the portion of the audio signal. The present disclosure further relates to a method of detecting dense transient events in a portion of an audio signal.

8. 发明申请

WO2015036352A1 CODING OF MULTICHANNEL AUDIO CONTENT 审中-公开
标题翻译：多通道音频内容的编码
公开(公告)号：WO2015036352A1
公开(公告)日：2015-03-19
申请号：PCT/EP2014/069044
申请日：2014-09-08
申请人： DOLBY INTERNATIONAL AB
发明人： PURNHAGEN, Heiko , MUNDT, Harald , KJOERLING, Kristofer
IPC分类号： G10L19/008
CPC分类号： G10L19/008 , G10L19/02 , G10L19/24 , H04S5/00 , H04S2400/03 , H04S2420/03
摘要： There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.
摘要翻译：提供了用于在具有N个通道的扬声器配置上播放的多声道音频内容的编码和解码的解码和编码方法。解码方法包括在第一解码模块中将M个输入音频信号解码为适合于在具有M个声道的扬声器配置上重放的M个mid信号; 并且对于超过M个通道的N个通道中的每一个，接收对应于M个中间信号之一的附加输入音频信号，并解码输入音频信号及其对应的中间信号，以便产生包括第一和第二信号的立体声信号第二音频信号适于在扬声器配置的N个通道中的两个上播放。

9. 发明申请

WO2020039000A1 CODING DENSE TRANSIENT EVENTS WITH COMPANDING 审中-公开
公开(公告)号：WO2020039000A1
公开(公告)日：2020-02-27
申请号：PCT/EP2019/072377
申请日：2019-08-21
申请人： DOLBY INTERNATIONAL AB
发明人： BISWAS, Arijit , MUNDT, Harald
IPC分类号： G10L21/034 , H04B1/64
摘要： Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A method of processing an audio signal includes the following operations. A system receives an audio signal. The system determines that a first frame of the audio signal includes a sparse transient signal. The system determines that a second frame of the audio signal includes a dense transient signal. The system compresses/expands (compands) the audio signal using a companding rule that applies a first companding exponent to the first frame of the audio signal and applies a second companding exponent to the second frame of the audio signal, each companding exponent being used to derive a respective degree of dynamic range compression and expansion for a corresponding frame. The system then provides the companded audio signal to a downstream device.

10. 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。

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