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21. 发明授权

EP2545646B1 SYSTEM FOR COMBINING LOUDNESS MEASUREMENTS IN A SINGLE PLAYBACK MODE 有权转让
标题翻译：用于在单个回放模式中组合回路测量的系统
公开(公告)号：EP2545646B1
公开(公告)日：2017-05-17
申请号：EP11707167.0
申请日：2011-03-07
申请人： Dolby International AB , Dolby Laboratories Licensing Corporation
发明人： WOLTERS, Martin , SCHUG, Michael , MUNDT, Harald, H. , RIEDMILLER, Jeffrey, C.
IPC分类号： H03G9/00 , H03G9/14
CPC分类号： H03G9/00 , H03G9/005 , H03G9/14

22. 发明授权

EP2981963B1 COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 有权转让
标题翻译： KOMPANDIERUNGSVORRICHTUNG和减少量化晚期频谱扩展程序
公开(公告)号：EP2981963B1
公开(公告)日：2017-01-04
申请号：EP14720877.1
申请日：2014-04-01
申请人： Dolby Laboratories Licensing Corporation , Dolby International AB
发明人： HEDELIN, Per , BISWAS, Arijit , SCHUG, Michael , MELKOTE, Vinay
IPC分类号： G10L21/034 , H03G3/24 , G03G7/00 , H04B1/66 , G10L25/18 , G10L25/45 , H03G7/00
CPC分类号： G10L21/0232 , G10L19/008 , G10L19/0208 , G10L19/032 , G10L21/034 , G10L25/18 , G10L25/45 , H03G7/007 , H04B1/66
摘要： Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

23. 发明公开

EP2917909A2 REDUCED COMPLEXITY CONVERTER SNR CALCULATION 有权转让
标题翻译： SNR-BERECHNUNG EINES WANDLERS MIT REDUZIERTERKOMPLEXITÄT
公开(公告)号：EP2917909A2
公开(公告)日：2015-09-16
申请号：EP13785889.0
申请日：2013-11-04
申请人： Dolby International AB , Dolby Laboratories Licensing Corporation
发明人： SCHUG, Michael , WILLIAMS, Phillip
IPC分类号： G10L19/032
CPC分类号： G10L19/008 , G10L19/02 , G10L19/032 , G10L19/173
摘要： The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for reducing the complexity of a bit allocation process used in the context of audio encoding/decoding. An audio encoder (300) configured to encode an audio signal according to a first audio codec system is described. The audio encoder (300) comprises a transform unit (302) configured to determine a set of spectral coefficients (312) based on the audio signal. Furthermore, the encoder (300) comprises a floating-point encoding unit (304) configured to determine a set of scale factors and a set of scaled values (314), based on the set of spectral coefficients (312); and to encode the set of scale factors to yield a set of encoded scale factors (313).
摘要翻译：本文件涉及音频编码/解码。特别地，本文件涉及用于降低在音频编码/解码的上下文中使用的比特分配过程的复杂度的方法和系统。描述了被配置为根据第一音频编解码器系统对音频信号进行编码的音频编码器（300）。音频编码器（300）包括被配置为基于音频信号确定一组频谱系数（312）的变换单元（302）。此外，编码器（300）包括浮点编码单元（304），其被配置为基于频谱系数集合（312）确定一组比例因子和一组缩放值（314）; 并编码一组比例因子以产生一组经编码的比例因子（313）。

24. 发明申请

WO2013124445A2 METHODS AND SYSTEMS FOR EFFICIENT RECOVERY OF HIGH FREQUENCY AUDIO CONTENT 审中-公开
标题翻译：有效恢复高频音频内容的方法和系统
公开(公告)号：WO2013124445A2
公开(公告)日：2013-08-29
申请号：PCT/EP2013/053609
申请日：2013-02-22
申请人： DOLBY INTERNATIONAL AB
发明人： THESING, Robin , SCHUG, Michael
IPC分类号： G10L21/0388
CPC分类号： G10L19/008 , G10L19/0204 , G10L19/028 , G10L19/167 , G10L21/0388
摘要： The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value (311, 312) for a first frequency subband (205) of an audio signal is described. The first banded tonality value (311, 312) is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values (341 ) for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values (341) for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value (311, 312) for the first frequency subband.
摘要翻译：本文件涉及音频编码，解码和处理的技术领域。具体涉及以有效的方式从同一音频信号的低频内容恢复音频信号的高频内容的方法。描述了一种用于确定音频信号的第一频率子带（205）的第一带状音调值（311,312）的方法。第一带状音调值（311,312）用于基于音频信号的低频分量来近似音频信号的高频分量。该方法包括基于音频信号的采样块确定对应的一组频率仓中的一组变换系数; 使用该组变换系数分别确定该组频率仓的一组仓
单调值（341）以及对位于所述第一频率子带内的所述频率仓的集合中的两个或更多个相应的相邻频率仓组合所述一组仓值音调值（341）中的两个或更多个的第一子集，由此产生所述第一带状音调值（311， 312）用于第一频率子带。

25. 发明申请

WO2011015369A1 AUTHENTICATION OF DATA STREAMS 审中-公开
标题翻译：数据流的验证
公开(公告)号：WO2011015369A1
公开(公告)日：2011-02-10
申请号：PCT/EP2010/004827
申请日：2010-08-06
申请人： DOLBY INTERNATIONAL AB , BOEHM, Reinhold , GROESCHEL, Alexander , HOERICH, Holger , HOMM, Daniel , SCHILDBACH, Wolfgang A. , SCHUG, Michael , WATZKE, Oliver , WOLTERS, Martin , ZIEGLER, Thomas
发明人： BOEHM, Reinhold , GROESCHEL, Alexander , HOERICH, Holger , HOMM, Daniel , SCHILDBACH, Wolfgang A. , SCHUG, Michael , WATZKE, Oliver , WOLTERS, Martin , ZIEGLER, Thomas
IPC分类号： G11B20/00 , G06F21/00 , H04L29/06 , H04N7/167 , H04N7/24 , H04N7/52 , H04H60/56
CPC分类号： G11B20/00086 , G11B20/00007 , G11B20/00166 , G11B20/00173 , G11B20/00188 , G11B20/0021 , G11B2020/00028 , H04H20/31 , H04H60/37 , H04H60/73 , H04H2201/50 , H04L65/602 , H04N21/23424 , H04N21/235 , H04N21/2541 , H04N21/435 , H04N21/44016 , H04N21/462 , H04N21/6332 , H04N21/654 , H04N21/8106 , H04N21/835
摘要： The present invention relates to techniques for authentication of data streams. Specifically, the invention relates to the insertion of identifiers into a data stream, such as a Dolby Pulse, AAC or HE AAC bitstream, and the authentication and verification of the data stream based on such identifiers. A method and system for encoding a data stream comprising a plurality of data frames is described. The method comprises the step of generating a cryptographic value of a number N of successive data frames and configuration information, wherein the configuration information comprises information for rendering the data stream. The method then inserts the cryptographic value into the data stream subsequent to the N successive data frames.
摘要翻译：本发明涉及用于认证数据流的技术。具体地，本发明涉及将标识符插入诸如杜比脉冲，AAC或HE AAC比特流之类的数据流，以及基于这些标识符的数据流的认证和验证。描述了一种用于对包括多个数据帧的数据流进行编码的方法和系统。该方法包括生成连续数据帧的N个密码值和配置信息的步骤，其中配置信息包括用于呈现数据流的信息。然后，该方法在N个连续数据帧之后将密码值插入到数据流中。

26. 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.

27. 发明申请

WO2013079524A2 ENHANCED CHROMA EXTRACTION FROM AN AUDIO CODEC 审中-公开
标题翻译：增强了从音频编解码器中提取色彩
公开(公告)号：WO2013079524A2
公开(公告)日：2013-06-06
申请号：PCT/EP2012/073825
申请日：2012-11-28
申请人： DOLBY INTERNATIONAL AB
发明人： BISWAS, Arijit , FINK, Marco , SCHUG, Michael
IPC分类号： G10L25/48
CPC分类号： G10L19/02 , G10H1/0008 , G10H1/383 , G10H2210/066 , G10H2250/225 , G10L19/022 , G10L19/038 , G10L21/0388 , G10L25/54
摘要： The present document relates to methods and systems for music information retrieval (MIR). In particular, the present document relates to methods and systems for extracting a chroma vector from an audio signal. A method (900) for determining a chroma vector (100) for a block of samples of an audio signal (301) is described. The method (900) comprises receiving (901) a corresponding block of frequency coefficients derived from the block of samples of the audio signal (301) from a core encoder (412) of a spectral band replication based audio encoder (410) adapted to generate an encoded bitstream (305) of the audio signal (301) from the block of frequency coefficients; and determining (904) the chroma vector (100) for the block of samples of the audio signal (301) based on the received block of frequency coefficients.
摘要翻译：本文件涉及用于音乐信息检索（MIR）的方法和系统。特别地，本文件涉及用于从音频信号中提取色度矢量的方法和系统。描述了用于确定音频信号（301）的样本块的色度矢量（100）的方法（900）。所述方法（900）包括从适于生成基于频谱复制的音频编码器（410）的核心编码器（412）接收（901）从所述音频信号（301）的采样块导出的频率系数块来自频率系数块的音频信号（301）的编码比特流（305）以及基于所接收的频率系数块来确定（904）音频信号（301）的样本块的色度向量（100）。

28. 发明申请

WO2012152764A1 METHOD AND ENCODER FOR PROCESSING A DIGITAL STEREO AUDIO SIGNAL 审中-公开
标题翻译：用于处理数字立体声音频信号的方法和编码器
公开(公告)号：WO2012152764A1
公开(公告)日：2012-11-15
申请号：PCT/EP2012/058391
申请日：2012-05-07
申请人： DOLBY INTERNATIONAL AB , SCHUG, Michael , MUNDT, Harald
发明人： SCHUG, Michael , MUNDT, Harald
IPC分类号： G10L19/00 , G10L19/02
CPC分类号： H04S1/007 , G10L19/008 , G10L19/03
摘要： The invention discloses a method and an encoder for processing a digital audio stereo signal. A digital audio encoder for coding such audio signal comprises a predictive Temporal Noise Shaping (TNS) filter, a Mid-/Side (M/S) coding unit, a control unit for determining a first prediction gain related to the unmodified L/R signal processed by the TNS filter and for determining a second prediction gain related to the M/S-coded L/R signal processed by the TNS filter, wherein the control unit is adapted to disable TNS-filtering - i.e. to bypass the TNS filter - for a current signal frame, if the first and second prediction gains differ by more than a pre-determined mismatch range. Preferably, the first and second prediction gains are determined from signal energy ratios calculated for each channel of the stereo signal including the signal energies of both the TNS-processed (unmodified) L- respectively (unmodified) R-signal and the TNS-processed M/S coded L- respectively M/S coded R-signal divided by the respective signal energies before TNS processing. Furthermore, the control unit is preferably adapted to overrule the disabling of the TNS filter, if the input signal is a near-mono audio signal exhibiting only low energy either in its M- or S-band. In that case, operation of the TNS filter on the stereo audio signal is maintained.
摘要翻译：本发明公开了一种用于处理数字音频立体声信号的方法和编码器。用于对这种音频信号进行编码的数字音频编码器包括预测时间噪声整形（TNS）滤波器，中/侧（M / S）编码单元，用于确定与未修改的L / R信号相关的第一预测增益的控制单元由TNS滤波器处理并确定与由TNS滤波器处理的M / S编码的L / R信号相关的第二预测增益，其中该控制单元用于禁用TNS滤波，即绕过TNS滤波器，以用于如果第一和第二预测增益差超过预定的失配范围，则当前信号帧。优选地，第一和第二预测增益是根据对包括TNS处理（未修改）L信号和TNS处理的M信号的两个信号能量的立体声信号的每个信道计算的信号能量比确定的 / S编码的L-分别M / S编码的R信号除以TNS处理之前的各个信号能量。此外，如果输入信号是在其M波段或S波段中仅表现出低能量的近乎单声道的音频信号，则控制单元优选地适用于推翻TNS滤波器的禁用。在这种情况下，维持TNS滤波器对立体声音频信号的操作。

29. 发明申请

WO2011051279A1 COMPLEXITY SCALABLE PERCEPTUAL TEMPO ESTIMATION 审中-公开
标题翻译：复杂可伸缩的概率估计
公开(公告)号：WO2011051279A1
公开(公告)日：2011-05-05
申请号：PCT/EP2010/066151
申请日：2010-10-26
申请人： DOLBY INTERNATIONAL AB , BISWAS, Arijit , HOLLOSI, Danilo , SCHUG, Michael
发明人： BISWAS, Arijit , HOLLOSI, Danilo , SCHUG, Michael
IPC分类号： G10H1/40
CPC分类号： G10H1/40 , G10H2210/076 , G10H2230/015 , G10H2240/075
摘要： The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit- stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity.
摘要翻译：本文件涉及用于估计媒体信号（诸如音频或组合视频/音频信号）的速度的方法和系统。特别地，该文件涉及人类听众感知的节奏的估计，以及用于以可缩放的计算复杂度进行速度估计的方法和系统。描述用于从包括频谱带复制数据的音频信号的编码比特流中提取音频信号的速度信息的方法和系统。该方法包括以下步骤：在音频信号的时间间隔中确定与包含在编码比特流中的频谱带复制数据量相关联的有效载荷数量; 重复该音频信号的编码比特流的连续时间间隔的确定步骤，从而确定有效载荷量的序列; 识别有效载荷数量序列中的周期; 以及从所识别的周期中提取音频信号的速度信息。

30. 发明申请

WO2022008571A2 PACKET LOSS CONCEALMENT 审中-公开
公开(公告)号：WO2022008571A2
公开(公告)日：2022-01-13
申请号：PCT/EP2021/068774
申请日：2021-07-07
申请人： DOLBY INTERNATIONAL AB
发明人： MUNDT, Harald , BRUHN, Stefan , PURNHAGEN, Heiko , PLAIN, Simon , SCHUG, Michael
IPC分类号： G10L19/005 , G10L19/008
摘要： Described are methods of processing an audio signal for packet loss concealment. The audio signal comprises a sequence of frames, each frame containing representations of a plurality of audio channels and reconstruction parameters for upmixing the plurality of audio channels to a predetermined channel format. One method includes: receiving the audio signal; and generating a reconstructed audio signal in the predefined channel format based on the received audio signal. Generating the reconstructed audio signal comprises: determining whether at least one frame of the audio signal has been lost; and if a number of consecutively lost frames exceeds a first threshold, fading the reconstructed audio signal to a predefined spatial configuration. Also described is a method of encoding an audio signal. Yet further described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

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