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1. 发明申请

US20140240447A1 Layered Mixing for Sound Field Conferencing System 有权
标题翻译：声场会议系统分层混音
公开(公告)号：US20140240447A1
公开(公告)日：2014-08-28
申请号：US14166065
申请日：2014-01-28
申请人： Dolby International AB , Dolby Laboratories Licensing Corporation
发明人： Richard J. Cartwright , Craig Johnston , Glenn N. Dickins , Heiko Purnhagen
IPC分类号： H04N7/15
CPC分类号： H04N7/152 , G10L19/008 , G10L25/78 , H04M3/567 , H04M3/569
摘要： A conferencing server (100) receives incoming bitstreams (I1, I2, I3, I4, I5) carrying media data from respective conferencing endpoints (110, 120, 130, 140, 150); receives a mixing strategy (M) specifying properties of at least one outgoing bitstream (O1, O2, O3, O4, O5) and requiring at least one additive media mixing step; and supplies at least one outgoing bitstream by executing, in a processor (103) and a memory (102) with a plurality of memory spaces, a run list of operations selected from a predefined collection of primitives and realizing the received mixing strategy. A pre-processor (104) in the server derives said run list repeatedly and dynamically while taking into consideration determined momentary activity in each incoming bitstream. In embodiments, the run list may be derived by (a) pruning of an initial run list, (b) constrained or non-constrained minimization of a cost function, or (c) automatic code generation.
摘要翻译：会议服务器（100）从相应的会议端点（110,120,130,140,150）接收携带媒体数据的传入比特流（I1，I2，I3，I4，I5）; 接收指定至少一个传出比特流（O1，O2，O3，O4，O5）的属性并需要至少一个添加介质混合步骤的混合策略（M）并且通过在具有多个存储器空间的处理器（103）和存储器（102）中执行从预定义的图元集合中选择的操作的运行列表并实现所接收的混合策略来提供至少一个输出比特流。服务器中的预处理器（104）在考虑每个输入比特流中确定的瞬时活动的同时，重复地和动态地导出所述运行列表。在实施例中，可以通过（a）修剪初始运行列表，（b）成本函数的约束或非约束最小化，或（c）自动代码生成来导出运行列表。

2. 发明授权

US09530422B2 Bitstream syntax for spatial voice coding 有权
标题翻译：空间语音编码的位流语法
公开(公告)号：US09530422B2
公开(公告)日：2016-12-27
申请号：US14392287
申请日：2014-06-26
申请人： Dolby Laboratories Licensing Corporation , DOLBY INTERNATIONAL AB
发明人： Janusz Klejsa , Leif Jonas Samuelsson , Heiko Purnhagen , Glenn N. Dickins
IPC分类号： G10L19/00 , G10L19/008 , G10L19/032 , G10L19/002 , G10L19/02 , G10L19/035
CPC分类号： G10L19/008 , G10L19/002 , G10L19/0204 , G10L19/0212 , G10L19/032 , G10L19/035
摘要： An encoding system (100) encodes a first (E1) and further (E2, E3) audio signals as a layered bitstream (B), wherein a quantizer for each frequency band of each signal is selected using a rate allocation rule based on signal-specific rate allocation data, a spectral envelope of the signal and a reference level (EnvE1Max), which is determined based on the spectral envelope of the first signal and is not necessarily included in the bitstream. Further disclosed is a decoding system for reconstructing the audio signals based on the bitstream. In embodiments, the bitstream has a basic layer (BE1), which contains data that enable decoding of the first audio signal, and a spatial layer (Bspatial) facilitating decoding of the further audio signal(s). In embodiments, the encoding system prepares the bitstream subject to a basic-layer bitrate constraint and a total bitrate constraint.
摘要翻译：编码系统（100）将第一（E1）和另外（E2，E3）音频信号编码为分层比特流（B），其中使用基于信号的比特率的速率分配规则来选择每个信号的每个频带的量化器，特定速率分配数据，信号的频谱包络和基于第一信号的频谱包络确定的参考电平（EnvE1Max），并且不一定包括在比特流中。还公开了一种用于基于比特流重建音频信号的解码系统。在实施例中，比特流具有包含能够对第一音频信号进行解码的数据的基本层（BE1）以及便于对其它音频信号进行解码的空间层（B空间）。在实施例中，编码系统根据基本层比特率约束和总比特率约束准备比特流。

3. 发明申请

US20150237301A1 NEAR-END INDICATION THAT THE END OF SPEECH IS RECEIVED BY THE FAR END IN AN AUDIO OR VIDEO CONFERENCE 有权
标题翻译：在音频或视频会议末尾接收到的语音结束的最终指示
公开(公告)号：US20150237301A1
公开(公告)日：2015-08-20
申请号：US14426134
申请日：2013-09-27
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： Dong Shi , Xuejing Sun , Kai Li , Shen Huang , Harald Mundt , Heiko Purnhagen , Glenn N. Dickins
IPC分类号： H04N7/14 , H04N7/15
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。估计器估计在远端的用户基于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。可感知的信号有助于避免各方之间的冲突。

4. 发明申请

US20220337969A1 ADAPTABLE SPATIAL AUDIO PLAYBACK 有权
公开(公告)号：US20220337969A1
公开(公告)日：2022-10-20
申请号：US17630098
申请日：2020-07-16
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： Alan J. Seefeldt , Joshua B. Lando , Daniel Arteaga , Glenn N. Dickins , Mark Richard Paul Thomas
IPC分类号： H04S7/00 , H04R5/02
摘要： A rendering mode may be determined for received audio data, including audio signals and associated spatial data. The audio data may be rendered for reproduction via a set of loudspeakers of an environment according to the rendering mode, to produce rendered audio signals. Rendering the audio data may involve determining relative activation of a set of loudspeakers in an environment. The rendering mode may be variable between a reference spatial mode and one or more distributed spatial modes. The reference spatial mode may have an assumed listening position and orientation. In the distributed spatial mode(s), one or more elements of the audio data may each be rendered in a more spatially distributed manner than in the reference spatial mode and spatial locations of remaining elements of the audio data may be warped such that they span a rendering space of the environment more completely than in the reference spatial mode.

5. 发明授权

US09905236B2 Enabling sampling rate diversity in a voice communication system 有权
公开(公告)号：US09905236B2
公开(公告)日：2018-02-27
申请号：US14384350
申请日：2013-03-21
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING CORPORATION
发明人： Heiko Purnhagen , Leif Sehlstrom , Lars Villemoes , Glenn N. Dickins , Mark S. Vinton
IPC分类号： G10L19/02 , G10L19/03 , H04M3/56 , G10L19/16 , G10L19/002 , G10L19/26
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 neighborhood 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.

6. 发明授权

US09502046B2 Coding of a sound field signal 有权
标题翻译：声场信号的编码
公开(公告)号：US09502046B2
公开(公告)日：2016-11-22
申请号：US14421132
申请日：2013-09-20
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： Heiko Purnhagen , Leif Jonas Samuelsson , Janusz Klejsa , Lars Villemoes , Toni Hirvonen , Glenn N. Dickins , Richard James Cartwright
IPC分类号： G10L19/012 , G10L19/008 , H04M3/56 , G10L21/0208 , G10L19/032 , G10L19/02 , G10L21/0216
CPC分类号： G10L19/012 , G10L19/008 , G10L19/02 , G10L19/0208 , G10L19/032 , G10L19/22 , G10L19/24 , G10L21/02 , G10L21/0208 , G10L21/0216 , H04M3/56
摘要： A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.
摘要翻译：用于编码声场信号的方法包括：通过在联合处理中对所有信号的所有子带应用均匀标准来分配编码速率。分配标准可以基于在给定子带中在待编码的信号的频谱包络与编码噪声分布之间的比较，其中噪声分布可以是噪声形状和噪声偏移的和，哪个噪声基于编码比特预算来计算偏移量。速率分配过程可以与能量压缩正交变换组合，其中提出了易于有效编码并且具有可调方向性的参数化。另一方面，本发明提供了相应的解码方法。

7. 发明授权

US10224046B2 Spatial comfort noise 有权
公开(公告)号：US10224046B2
公开(公告)日：2019-03-05
申请号：US14774966
申请日：2014-03-04
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： Glenn N. Dickins , Xuejing Sun , Yen-Liang Shue , Heiko Purnhagen
IPC分类号： G10L19/012 , H04L12/18 , G10L21/0208 , H04M3/56 , G10L19/008
摘要： A method, an apparatus, logic (e.g., executable instructions encoded in a non-transitory computer-readable medium to carry out a method), and a non-transitory computer-readable medium configured with such instructions. The method is to generate and spatially render spatial comfort noise at a receiving endpoint of a conference system, such that the comfort noise has target spectral characteristics typical of comfort noise, and at least one spatial property that at least substantially matches at least one target spatial property. On version includes receiving one or more or more audio signals from other endpoints, combining the received audio signals with the spatial comfort noise signals, and rendering the combination of the received audio signals and the spatial comfort noise signals to a set of output signals for loudspeakers, such that the spatial comfort noise signals are continually in the output signal sin addition to output from the received audio signals.

8. 发明申请

US20150221313A1 CODING OF A SOUND FIELD SIGNAL 有权
标题翻译：声场信号的编码
公开(公告)号：US20150221313A1
公开(公告)日：2015-08-06
申请号：US14421132
申请日：2013-09-20
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： Heiko Purnhagen , Leif Jonas Samuelsson , Janusz Klejsa , Lars Villemoes , Toni Hirvonen , Glenn N. Dickins , Richard James Cartwright
IPC分类号： G10L19/008 , G10L19/02
CPC分类号： G10L19/012 , G10L19/008 , G10L19/02 , G10L19/0208 , G10L19/032 , G10L19/22 , G10L19/24 , G10L21/02 , G10L21/0208 , G10L21/0216 , H04M3/56
摘要： A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.
摘要翻译：用于编码声场信号的方法包括：通过在联合处理中对所有信号的所有子带应用均匀标准来分配编码速率。分配标准可以基于在给定子带中在待编码的信号的频谱包络与编码噪声分布之间的比较，其中噪声分布可以是噪声形状和噪声偏移的和，哪个噪声基于编码比特预算来计算偏移量。速率分配过程可以与能量压缩正交变换组合，其中提出了易于有效编码并且具有可调方向性的参数化。另一方面，本发明提供了相应的解码方法。

9. 发明申请

US20150025896A1 Enabling Sampling Rate Diversity In A Voice Communication System 有权
标题翻译：在语音通信系统中实现采样率分集
公开(公告)号：US20150025896A1
公开(公告)日：2015-01-22
申请号：US14384350
申请日：2013-03-21
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING
发明人： Heiko Purnhagen , Leif Sehlstrom , Lars Villemoes , Glenn N. Dickins , Mark S. Vinton
IPC分类号： G10L19/03 , G10L19/26 , G10L19/002
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裂频率之间的范围有关的中性值频谱分量，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。

10. 发明授权

US11587576B2 Background noise estimation using gap confidence 有权
公开(公告)号：US11587576B2
公开(公告)日：2023-02-21
申请号：US17449918
申请日：2021-10-04
申请人： Dolby Laboratories Licensing Corporation
发明人： Christopher Graham Hines , Glenn N. Dickins , Adam J. Mills
IPC分类号： G10L21/0232 , H04R1/08 , G10L21/0208 , G10L21/0216 , H04R3/02
摘要： A noise estimation method including steps of generating gap confidence values in response to microphone output and playback signals, and using the gap confidence values to generate an estimate of background noise in a playback environment. Each gap confidence value is indicative of confidence of presence of a gap at a corresponding time in the playback signal, and may be a combination of candidate noise estimates weighted by the gap confidence values. Generation of the candidate noise estimates may but need not include performance of echo cancellation. Optionally, noise compensation is performed on an audio input signal using the generated background noise estimate. Other aspects are systems configured to perform any embodiment of the noise estimation method.

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