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1. 发明申请

WO2022216542A1 MULTI-BAND DUCKING OF AUDIO SIGNALS TECHNICAL FIELD 审中-公开
公开(公告)号：WO2022216542A1
公开(公告)日：2022-10-13
申请号：PCT/US2022/023057
申请日：2022-04-01
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： TYAGI, Rishabh , PURNHAGEN, Heiko
IPC分类号： G10L19/008
摘要： A method for multi-band ducking of audio signals is provided. In some implementations, the method involves receiving, at a decoder, an input audio signal, wherein the input audio signal is a downmixed audio signal. In some implementations, the method involves separating the input audio signal into a first set of frequency bands. In some implementations, the method involves determining a set of ducking gains, a ducking gain corresponding to a frequency band of the first set of frequency bands. In some implementations, the method involves generating a broadband decorrelated audio signal, wherein ducking gains of the set of ducking gains are applied to at least one of: 1) a second set of frequency bands prior to generating the at least one broadband decorrelated audio signal; or 2) a third set of frequency bands that separates the at least one broadband decorrelated audio signal.

2. 发明申请

WO2018175347A1 BACKWARD-COMPATIBLE INTEGRATION OF HARMONIC TRANSPOSER FOR HIGH FREQUENCY RECONSTRUCTION OF AUDIO SIGNALS 审中-公开
公开(公告)号：WO2018175347A1
公开(公告)日：2018-09-27
申请号：PCT/US2018/023183
申请日：2018-03-19
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING CORPORATION
发明人： VILLEMOES, Lars , PURNHAGEN, Heiko , EKSTRAND, Per
IPC分类号： G06F17/10 , G10L19/02 , G10L19/22 , G10L19/24 , G10L19/26
摘要： A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

3. 发明申请

WO2014210284A1 BITSTREAM SYNTAX FOR SPATIAL VOICE CODING 审中-公开
标题翻译：用于空间语音编码的BITSTREAM语法
公开(公告)号：WO2014210284A1
公开(公告)日：2014-12-31
申请号：PCT/US2014/044295
申请日：2014-06-26
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： KLEJSA, Janusz , SAMUELSSON, Jonas , PURNHAGEN, Heiko , DICKINS, Glenn N.
IPC分类号： G10L19/032 , G10L19/008
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 (B E1 ), which contains data that enable decoding of the first audio signal, and a spatial layer (B spatial ) 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并且不一定包括在比特流中的参考电平（EnvE1Max）。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。

4. 发明申请

WO2020010072A1 METHODS AND DEVICES FOR ENCODING AND/OR DECODING IMMERSIVE AUDIO SIGNALS 审中-公开
公开(公告)号：WO2020010072A1
公开(公告)日：2020-01-09
申请号：PCT/US2019/040282
申请日：2019-07-02
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： MCGRATH, David S. , ECKERT, Michael , PURNHAGEN, Heiko , BRUHN, Stefan
IPC分类号： G10L19/008 , G10L19/02 , G10L19/04 , G10L19/18
摘要： The present document describes a method (700) for encoding a multi-channel input signal (201). The method (700) comprises determining (701) a plurality of downmix channel signals (203) from the multi-channel input signal (201) and performing (702) energy compaction of the plurality of downmix channel signals (203) to provide a plurality of compacted channel signals (404). Furthermore, the method (700) comprises determining (703) joint coding metadata (205) based on the plurality of compacted channel signals (404) and based on the multi-channel input signal (201), wherein the joint coding metadata (205) is such that it allows upmixing of the plurality of compacted channel signals (404) to an approximation of the multi-channel input signal (201). In addition, the method (700) comprises encoding (704) the plurality of compacted channel signals (404) and the joint coding metadata (205).

5. 发明申请

WO2016040623A1 RENDERING AUDIO OBJECTS IN A REPRODUCTION ENVIRONMENT THAT INCLUDES SURROUND AND/OR HEIGHT SPEAKERS 审中-公开
标题翻译：在包括环绕和/或高度扬声器的复制环境中渲染音频对象
公开(公告)号：WO2016040623A1
公开(公告)日：2016-03-17
申请号：PCT/US2015/049416
申请日：2015-09-10
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： BREEBAART, Dirk Jeroen , SOLE, Antonio Mateos , PURNHAGEN, Heiko , TSINGOS, Nicolas R.
IPC分类号： H04S7/00 , H04R3/12
CPC分类号： H04S7/30 , H04R3/12 , H04R2400/11 , H04S2400/03 , H04S2400/11
摘要： During a process, decorrelation may be selectively applied to audio data for an audio object based, at least in part, on whether a speaker for which speaker feed signals will be determined is a surround speaker. In some implementations, decorrelation may be selectively applied according to whether such a speaker is a height speaker. Some implementations may reduce, or even eliminate, audio artifacts such as comb-filter notches and peaks. Some such implementations may increase the size of a "sweet spot" of a reproduction environment.
摘要翻译：在处理过程中，至少部分地，至少部分地基于是否将确定扬声器馈送信号的扬声器是环绕扬声器，可以将解相关选择性地应用于音频对象的音频数据。在一些实现中，可以根据这样的扬声器是高度扬声器来选择性地应用去相关。一些实现可以减少甚至消除诸如梳状滤波器缺口和峰值的音频伪影。一些这样的实现可以增加再现环境的“最佳点”的大小。

6. 发明申请

WO2015006112A1 PROCESSING OF TIME-VARYING METADATA FOR LOSSLESS RESAMPLING 审中-公开
标题翻译：用于无损修复的时变元件的处理
公开(公告)号：WO2015006112A1
公开(公告)日：2015-01-15
申请号：PCT/US2014/045156
申请日：2014-07-01
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： ARNOTT, Brian George , BREEBAART, Dirk Jeroen , MATEOS SOLE, Antonio , MCGRATH, David S. , PURNHAGEN, Heiko , SANCHEZ, Freddie , TSINGOS, Nicolas R.
IPC分类号： G10L19/008 , G10L19/16
CPC分类号： G10L19/0017 , G10L19/005 , G10L19/008 , G10L19/167 , G10L19/24
摘要： Embodiments are directed to a method of representing spatial rendering metadata for processing in an object-based audio system that allows for lossless interpolation and/or re-sampling of the metadata. The method comprises time stamping the metadata to create metadata instances, and encoding an interpolation duration to with each metadata instance that specifies the time to reach a desired rendering state for the respective metadata instance. The re-sampling of metadata is useful for re-clocking metadata to an audio coder and for the editing audio content.
摘要翻译：实施例涉及一种表示用于在基于对象的音频系统中进行处理的空间渲染元数据的方法，其允许元数据的无损内插和/或重新采样。该方法包括时间戳元数据以创建元数据实例，以及对每个元数据实例编码内插持续时间，该元数据实例指定为相应的元数据实例达到期望的呈现状态的时间。元数据的重新采样对于将元数据重新计时到音频编码器和编辑音频内容是有用的。

7. 发明申请

WO2014143582A1 SPATIAL COMFORT NOISE 审中-公开
标题翻译：空气舒适的噪音
公开(公告)号：WO2014143582A1
公开(公告)日：2014-09-18
申请号：PCT/US2014/020239
申请日：2014-03-04
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： DICKINS, Glenn N. , SUN, Xuejing , SHUE, Yen-Liang , PURNHAGEN, Heiko
IPC分类号： G10L19/012 , G10L19/008 , H04M3/56
CPC分类号： G10L19/012 , G10L19/008 , G10L21/0208 , H04L12/1813 , H04L12/1827 , H04M3/568
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.
摘要翻译：一种方法，装置，逻辑（例如，在非暂时性计算机可读介质中编码以执行方法的可执行指令）以及配置有这种指令的非暂时计算机可读介质。该方法是在会议系统的接收端产生和空间地呈现空间舒适噪声，使得舒适噪声具有典型的舒适噪声的目标频谱特性，以及至少基本匹配至少一个目标空间属性。 On版本包括从其他端点接收一个或多个或更多个音频信号，将接收到的音频信号与空间舒适噪声信号组合，以及将接收到的音频信号和空间舒适噪声信号的组合呈现给用于扬声器的一组输出信号使得空间舒适噪声信号连续地在输出信号sin中，从接收到的音频信号输出。

8. 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。

9. 发明申请

WO2014046923A1 AUDIO CODING WITH GAIN PROFILE EXTRACTION AND TRANSMISSION FOR SPEECH ENHANCEMENT AT THE DECODER 审中-公开
标题翻译：音频编码与增益简档提取和传输在解码器中进行语音增强
公开(公告)号：WO2014046923A1
公开(公告)日：2014-03-27
申请号：PCT/US2013/059144
申请日：2013-09-11
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： DICKINS, Glenn , PURNHAGEN, Heiko , SAMUELSSON, Leif J.
IPC分类号： G10L21/02 , H04M3/56 , H03G9/02 , 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： The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accord- ance with decomposition parameters controlling the quantitative properties of an or- thogonal energy-compacting transform, into rotated audio signals. Further, a time- variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also com- prise mutually independent broadband gains. Because signals in the audio coding format can be mixed with a limited computational effort, the invention may advanta- geously be applied in a tele-conferencing application.
摘要翻译：本发明提供了具有单声道层和至少一个声场层的分层音频编码格式。根据控制正交能量压缩变换的定量特性的分解参数，多个音频信号被分解成旋转的音频信号。此外，推导出时间变量增益曲线，其具体地指定如何处理旋转的音频信号以衰减不期望的音频内容。单声道层可以包括旋转的信号之一和增益分布。声场层可以包括旋转的信号和分解参数。在一个实施例中，增益分布包括清除增益曲线，其主要目的是消除非语音分量和/或噪声。增益曲线也可以相互独立的宽带增益。因为音频编码格式的信号可以以有限的计算量来混合，所以本发明可以有利地应用在电话会议应用中。

10. 发明申请

WO2014044812A1 CODING OF A SOUND FIELD SIGNAL 审中-公开
标题翻译：声场信号的编码
公开(公告)号：WO2014044812A1
公开(公告)日：2014-03-27
申请号：PCT/EP2013/069607
申请日：2013-09-20
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING CORPORATION
发明人： PURNHAGEN, Heiko , SAMUELSSON, Leif Jonas , KLEJSA, Janusz , VILLEMOES, Lars , HIRVONEN, Toni , DICKINS, Glenn N. , Richard James, Cartwright
IPC分类号： G10L19/008 , G10L19/02 , G10L19/032
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。

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