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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

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.

3. 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。

4. 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.

5. 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。

6. 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。

7. 发明申请

WO2022214480A1 ENCODING OF ENVELOPE INFORMATION OF AN AUDIO DOWNMIX SIGNAL 审中-公开
公开(公告)号：WO2022214480A1
公开(公告)日：2022-10-13
申请号：PCT/EP2022/059005
申请日：2022-04-05
申请人： DOLBY INTERNATIONAL AB
发明人： MUNDT, Harald
IPC分类号： G10L19/008 , G10L19/02
摘要： A method for encoding envelope information is provided. In some implementations, the method involves determining a first downmixed signal associated with a downmixed channel associated with an audio signal to be encoded. In some implementations, the method involves determining energy levels of the first downmixed signal for a plurality of frequency bands. In some implementations, the method involves determining whether to encode information indicative of the energy levels in a bitstream. In some implementations, the method involves encoding the determined energy levels. In some implementations, the method involves generating an energy control value indicating that energy levels are encoded. In some implementations, the method involves generating the bitstream, wherein the energy control value and the information indicative of the energy levels are usable by the decoder to adjust energy levels associated with the first downmixed signal.

8. 发明申请

WO2014161995A1 AUDIO ENCODER AND DECODER FOR INTERLEAVED WAVEFORM CODING 审中-公开
标题翻译：音频编码器和解码器用于互换波形编码
公开(公告)号：WO2014161995A1
公开(公告)日：2014-10-09
申请号：PCT/EP2014/056856
申请日：2014-04-04
申请人： DOLBY INTERNATIONAL AB
发明人： KJOERLING, Kristofer , THESING, Robin , MUNDT, Harald , PURNHAGEN, Heiko , ROEDEN, Karl Jonas
IPC分类号： G10L19/02 , G10L21/038
CPC分类号： G10L19/0208 , G10L19/02 , G10L19/0212 , G10L19/26 , G10L21/038 , G10L21/0388
摘要： There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.
摘要翻译：提供了用于对音频信号进行解码和编码的方法和装置。特别地，一种解码方法包括：接收波形编码信号，该波形编码信号具有对应于高于交叉频率的频率范围子集的频谱内容。波形编码信号与高于交叉频率的音频信号的参数高频重构进行交织。以这种方式，实现了音频信号的高频带的改进的重建。

9. 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。

10. 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.

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