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1. 发明专利

SG184815A1 AUDIO ENCODER, AUDIO DECODER AND RELATED METHODS FOR PROCESSING MULTI-CHANNEL AUDIO SIGNALS USING COMPLEX PREDICTION 未知
公开(公告)号：SG184815A1
公开(公告)日：2012-11-29
申请号：SG2012075214
申请日：2011-03-23
申请人： FRAUNHOFER GES DER ANGEWANDTEN FORSCHUNG E V , DOLBY INT AB
发明人： PURNHAGEN HEIKO , CARLSSON PONTUS , VILLEMOES LARS , ROBILLARD JULIEN , NEUSINGER MATTHIAS , HELMRICH CHRISTIAN , HILPERT JOHANNES , RETTELBACH NIKOLAUS , DISCH SASCHA , EDLER BERND
IPC分类号： G10L19/008 , G10L19/04
摘要： An audio encoder and an audio decoder are based on a combination of two audio channels (201, 202) to obtain a first combination signal (204) as a mid signal and a residual signal (205) which can be derived using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded (209) and written (212) into a data stream (213) together with the prediction information (206) derived by an optimizer (207) based on an optimization target (208). A decoder uses the prediction residual signal, the first combination signal and the prediction information to derive a decoded first channel signal and a decoded second channel signal. In an encoder example or in a decoder example, a real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

2. 发明专利

AU2011238010A1 Audio encoder, audio decoder and related methods for processing multi-channel audio signals using complex prediction 未知
公开(公告)号：AU2011238010A1
公开(公告)日：2012-11-08
申请号：AU2011238010
申请日：2011-03-23
申请人： DOLBY INT AB , FRAUNHOFER GES DER ANGEWANDTEN FORSCHUNG E V
发明人： PURNHAGEN HEIKO , CARLSSON PONTUS , VILLEMOES LARS , ROBILLARD JULIEN , NEUSINGER MATTHIAS , HELMRICH CHRISTIAN , HILPERT JOHANNES , RETTELBACH NIKOLAUS , DISCH SASCHA , EDLER BERND
IPC分类号： G10L19/00 , G10L19/008 , G10L19/04
摘要： An audio encoder and an audio decoder are based on a combination of two audio channels (201, 202) to obtain a first combination signal (204) as a mid signal and a residual signal (205) which can be derived using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded (209) and written (212) into a data stream (213) together with the prediction information (206) derived by an optimizer (207) based on an optimization target (208). A decoder uses the prediction residual signal, the first combination signal and the prediction information to derive a decoded first channel signal and a decoded second channel signal. In an encoder example or in a decoder example, a real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

3. 发明公开

KR20180071418A 적어도 하나의 필 요소 내의 향상된 스펙트럼 대역 복제 메타데이터를 사용한 오디오 비트스트림들의 디코딩 审中-公开
公开(公告)号：KR20180071418A
公开(公告)日：2018-06-27
申请号：KR20187017423
申请日：2016-03-10
申请人： DOLBY INT AB
发明人： VILLEMOES LARS , PURNHAGEN HEIKO , EKSTRAND PER
IPC分类号： G10L19/16 , G10L19/035 , G10L19/24 , G10L21/038
CPC分类号： G10L19/167 , G10L19/035 , G10L19/24 , G10L21/038
摘要： 실시예들은버퍼, 비트스트림페이로드디포맷터, 및디코딩서브시스템을포함하는오디오처리유닛에관한것이다. 버퍼는인코딩된오디오비트스트림의적어도하나의블록을저장한다. 블록은식별자로시작하여그 다음에필 데이터가오는필 요소를포함한다. 필데이터는향상된스펙트럼대역복제(eSBR) 처리가블록의오디오콘텐츠에대해수행되어야하는지를식별하는적어도하나의플래그를포함한다. 인코딩된오디오비트스트림을디코딩하기위한대응하는방법이또한제공된다.

4. 发明公开

KR20200137026A 후처리 지연을 저감시킨 고주파 재구성 기술의 통합 审中-公开
公开(公告)号：KR20200137026A
公开(公告)日：2020-12-08
申请号：KR20207033980
申请日：2019-04-25
申请人： DOLBY INT AB
发明人： KJOERLING KRISTOFER , VILLEMOES LARS , PURNHAGEN HEIKO , EKSTRAND PER
IPC分类号： G10L19/16 , G10L19/18 , G10L21/038
摘要： 인코딩된오디오비트스트림을디코딩하기위한방법이개시된다. 방법은인코딩된오디오비트스트림을수신하는것 및디코딩된저대역오디오신호를생성하기위해오디오데이터를디코딩하는것을포함한다. 방법은고주파재구성메타데이터를추출하는것 및필터링된저대역오디오신호를생성하기위해분석필터뱅크로디코딩된저대역오디오신호를필터링하는것을더 포함한다. 방법은또한, 오디오데이터에대해스펙트럼변환또는고조파전위가수행되어야하는지여부를나타내는플래그를추출하는것 및플래그에따라필터링된저대역오디오신호및 고주파재구성메타데이터를사용하여오디오신호의고대역부분을재생성하는것을포함한다. 고주파재생성은오디오채널마다 3010개샘플의지연으로후처리동작으로서수행된다.

5. 发明申请

WO2014161991A2 AUDIO ENCODER AND DECODER 审中-公开
标题翻译：音频编码器和解码器
公开(公告)号：WO2014161991A2
公开(公告)日：2014-10-09
申请号：PCT/EP2014056851
申请日：2014-04-04
申请人： DOLBY INT AB
发明人： VILLEMOES LARS , KLEJSA JANUSZ , HEDELIN PER
IPC分类号： G10L19/02
CPC分类号： G10L19/032 , G10L19/02 , G10L19/06
摘要： The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a transform-based audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder (100, 170) configured to encode a speech signal into a bitstream is described. The encoder (100, 170) comprises a framing unit (101) configured to receive a set(132, 332) of blocks; wherein the set (132, 332) of blocks comprises a plurality of sequential blocks (131) of transform coefficients; wherein the plurality of blocks (131) is indicative of samples of the speech signal; wherein a block (131) of transform coefficients comprises a plurality of transform coefficients for a corresponding plurality of frequency bins (301). Furthermore, the encoder (100, 170) comprises an envelope estimation unit (102) configured to determine a current envelope (133) based on the plurality of sequential blocks (131) of transform coefficients; wherein the current envelope (133) is indicative of a plurality of spectral energy values (303) for the corresponding plurality of frequency bins (301). In addition, the encoder (100, 170) comprises an envelope interpolation unit (104) configured to determine a plurality of interpolated envelopes (136) for the plurality of blocks (131) of transform coefficients, respectively, based on the current envelope (133); Furthermore, the encoder (100, 170) comprises a flattening unit (108) configured to determine a plurality of blocks (140) of flattened transform coefficients by flattening the corresponding plurality of blocks (131) of transform coefficients using the corresponding plurality of interpolated envelopes (136), respectively; wherein the bitstream is determined based on the plurality of blocks (140) of flattened transform coefficients.
摘要翻译：本文件涉及音频编码和解码系统（称为音频编解码器系统）。特别地，本文件涉及特别适合于语音编码/解码的基于变换的音频编解码器系统。描述了被配置为将语音信号编码为比特流的基于变换的语音编码器（100,170）。编码器（100,170）包括被配置为接收一组（132,332）块的成帧单元（101）其中所述块（132,332）包括变换系数的多个顺序块（131）; 其中所述多个块（131）指示所述语音信号的采样; 其中变换系数的块（131）包括用于对应的多个频率仓（301）的多个变换系数。此外，编码器（100,170）包括被配置为基于变换系数的多个顺序块（131）确定当前包络（133）的包络估计单元（102）其中所述电流包络（133）指示对应的多个频率仓（301）的多个频谱能量值（303）。另外，编码器（100,170）包括一个包络插值单元（104），被配置为基于当前的信封（133）确定多个块（131）的变换系数的多个内插包络（136））; 此外，编码器（100,170）包括平坦化单元（108），其被配置为通过使用对应的多个内插信封来平坦化变换系数的相应多个块（131）来确定平坦化变换系数的多个块（140）（136）; 其中，基于所述平坦化变换系数的所述多个块（140）来确定所述比特流。

6. 发明申请

WO2014161996A3 AUDIO PROCESSING SYSTEM 审中-公开
标题翻译：音频处理系统
公开(公告)号：WO2014161996A3
公开(公告)日：2014-12-04
申请号：PCT/EP2014056857
申请日：2014-04-04
申请人： DOLBY INT AB
发明人： KJOERLING KRISTOFER , PURNHAGEN HEIKO , VILLEMOES LARS
IPC分类号： G10L19/008 , G10L19/04 , G10L19/20
CPC分类号： G10L19/008 , G10L19/032 , G10L19/04 , G10L19/20
摘要： An audio processing system (100) comprises a front-end component (102, 103), which receives quantized spectral components and performs an inverse quantization, yielding a time-domain representation of an intermediate signal. The audio processing system further comprises a frequency-domain processing stage (104, 105, 106, 107, 108), configured to provide a time-domain representation of a processed audio signal, and a sample rate converter (109), providing a reconstructed audio signal sampled at a target sampling frequency. The respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the time-domain representation of the processed audio signal are equal. In particular embodiments, the processing stage comprises a parametric upmix stage which is operable in at least two different modes and is associated with a delay stage that ensures constant total delay.
摘要翻译：音频处理系统（100）包括前端组件（102,103），其接收量化的频谱分量并执行逆量化，产生中间信号的时域表示。音频处理系统还包括频域处理级（104,105,106,107,108），被配置为提供经处理的音频信号的时域表示，以及采样率转换器（109），提供重建的以目标采样频率采样的音频信号。中间音频信号的时域表示和经处理的音频信号的时域表示的各自的内部采样率相等。在特定实施例中，处理阶段包括可在至少两个不同模式下操作的参数上混级，并且与确保恒定总延迟的延迟级相关联。

7. 发明申请

WO2013186344A3 SMOOTH CONFIGURATION SWITCHING FOR MULTICHANNEL AUDIO RENDERING BASED ON A VARIABLE NUMBER OF RECEIVED CHANNELS 审中-公开
标题翻译：基于可变数量的接收频道进行多通道音频渲染的最佳配置切换
公开(公告)号：WO2013186344A3
公开(公告)日：2014-02-06
申请号：PCT/EP2013062340
申请日：2013-06-14
申请人： DOLBY INT AB
发明人： PURNHAGEN HEIKO , SEHLSTROM LEIF , ROEDEN KARL JONAS , KJOERLING KRISTOFER , VILLEMOES LARS
IPC分类号： G10L19/008 , G10L19/005
CPC分类号： G10L19/008 , G10L19/0017 , G10L19/18 , H04S3/008 , H04S2400/03 , H04S2420/03
摘要： A decoding system reconstructs an n-channel audio signal on the basis of an input signal representing the audio signal, in different time frames, either by parametric coding or as n discretely coded channels. Parametric decoding uses a core signal and mixing parameters controlling a spatial synthesis stage, to which a downmix signal is supplied from a downmix stage. The downmix stage realizes a projection on the downmix signal based on an n- channel input signal, either a discretely coded signal or a core signal padded with neutral-valued channels. The padding may take place either on the decoding side (reduced parametric coding) or the encoding side. In an embodiment, an audio decoder (110) in the decoding system pads the core signal during an initial portion of each reduced parametrically coded time frame directly succeeding a discretely coded time frame and during a final portion of each reduced parametrically coded time frame directly preceding a discretely coded time frame.
摘要翻译：解码系统通过参数编码或作为n个离散编码的信道，在不同的时间帧中，基于表示音频信号的输入信号来重构n信道音频信号。参数解码使用核心信号和控制空间合成阶段的混合参数，下混合信号从下混合级提供给它。下混合阶段基于n-沟道输入信号，即离散编码信号或填充有中性信道的核心信号实现对下混信号的投影。填充可以在解码侧（缩减参数编码）或编码侧进行。在一个实施例中，解码系统中的音频解码器（110）在直接在离散编码的时间帧之后的每个缩减的参数编码的时间帧的初始部分和在直接前进的每个缩减的参数编码的时间帧的最后部分期间，离散编码的时间帧。

8. 发明申请

WO2014187989A3 RECONSTRUCTION OF AUDIO SCENES FROM A DOWNMIX 审中-公开
标题翻译：从下载重建音频场景
公开(公告)号：WO2014187989A3
公开(公告)日：2015-02-19
申请号：PCT/EP2014060732
申请日：2014-05-23
申请人： DOLBY INT AB
发明人： HIRVONEN TONI , PURNHAGEN HEIKO , SAMUELSSON LEIF JONAS , VILLEMOES LARS
IPC分类号： G10L19/008 , H04S7/00
CPC分类号： G10L19/008 , G10L19/0204 , G10L19/20 , G10L25/06 , H04S3/008 , H04S3/02 , H04S5/00 , H04S7/30 , H04S2400/03 , H04S2400/11 , H04S2420/03
摘要： Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency- dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.
摘要翻译：音频对象与位置元数据相关联。接收的下混合信号包括作为一个或多个音频对象的线性组合的下混通道，并且与相应的位置定位器相关联。在第一方面，接收降混信号，位置元数据和与频率相关的对象增益。根据基于位置元数据和位置定位器的系数，将对象增益应用于缩混信号的上混合来重构音频对象。在第二方面，音频对象已经与位于对应的下混通道的位置定位器处的至少一个床通道一起被编码。解码系统接收下混合信号和音频对象的位置元数据。基于对应的下混通道的位置定位器，通过从对应的下混通道中抑制表示音频对象的内容来重构床通道。

9. 发明申请

WO2014161994A3 ADVANCED QUANTIZER 审中-公开
标题翻译：高级量器
公开(公告)号：WO2014161994A3
公开(公告)日：2014-11-27
申请号：PCT/EP2014056855
申请日：2014-04-04
申请人： DOLBY INT AB
发明人： KLEJSA JANUSZ , VILLEMOES LARS , HEDELIN PER
IPC分类号： G10L19/035
CPC分类号： G10L19/035 , G10L19/005 , G10L19/028 , G10L19/20
摘要： The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a transform-based audio codec system which is particularly well suited for voice encoding/decoding. A quantization unit (112) configured to quantize a first coefficient of a block (141) of coefficients is described. The block (141) of coefficients comprises a plurality of coefficients for a plurality of corresponding frequency bins (301). The quantization unit (112) is configured to provide a set (326, 327) of quantizers. The set (326, 327) of quantizers comprises a plurality of different quantizers (321, 322, 323) associated with a plurality of different signal-to-noise ratios, referred to as SNR, respectively. The plurality of different quantizers (321, 322, 323) includes a noise-filling quantizer (321); one or more dithered quantizers (322); and one or more un-dithered quantizers (323). The quantization unit (112) is further configured to determine an SNRindication indicative of a SNR attributed to the first coefficient, and to select a first quantizer from the set (326, 327) of quantizers, based on the SNR indication. In addition, the quantization unit (112) is configured to quantize the first coefficient using the first quantizer.
摘要翻译：本文件涉及音频编码和解码系统（称为音频编解码器系统）。特别地，本文件涉及特别适合于语音编码/解码的基于变换的音频编解码器系统。描述被配置为量化系数的块（141）的第一系数的量化单元（112）。系数块（141）包括用于多个相应频率仓（301）的多个系数。量化单元（112）被配置为提供量化器的集合（326,327）。量化器的集合（326,327）分别包括与多个不同的信噪比相关联的多个不同的量化器（321,322,323），分别称为SNR。多个不同的量化器（321,322,323）包括噪声填充量化器（321）; 一个或多个抖动量化器（322）; 和一个或多个未抖动量化器（323）。量化单元（112）还被配置为确定指示归因于第一系数的SNR的SNR指示，并且基于SNR指示从量化器的集合（326,327）中选择第一量化器。此外，量化单元（112）被配置为使用第一量化器来量化第一系数。

10. 发明申请

WO2013186343A3 SMOOTH CONFIGURATION SWITCHING FOR MULTICHANNEL AUDIO 审中-公开
标题翻译：多通道音频的最佳配置切换
公开(公告)号：WO2013186343A3
公开(公告)日：2014-02-06
申请号：PCT/EP2013062339
申请日：2013-06-14
申请人： DOLBY INT AB
发明人： PURNHAGEN HEIKO , SEHLSTROM LEIF , ROEDEN KARL JONAS , KJOERLING KRISTOFER , VILLEMOES LARS
IPC分类号： G10L19/18
CPC分类号： G10L19/008 , G10L19/0017 , G10L19/18 , H04S3/008 , H04S2400/03 , H04S2420/03
摘要： A decoding system (100) reconstructs an n-channel audio signal on the basis of an input signal (A) representing the audio signal either by parametric coding or as n discretely coded channels. Parametric decoding proceeds on the basis of a core signal and mixing parameters (a) controlling a spatial synthesis stage (150), which is supplied with a downmix signal from a downmix stage (140). A selector (170) controls the components of the decoding system, in steady-state parametric and discrete decoding mode and transitions between these. The downmix stage realizes a projection on the downmix signal based on an n-channel signal, either an n-channel input signal or a core signal padded with neutral values. The downmix stage is active in each time frame in which the input signal represents the audio signal by parametric coding and in at least the first time frame after the last time frame in each episode of parametrically coded time frames.
摘要翻译：解码系统（100）根据表示音频信号的输入信号（A），通过参数编码或作为n个离散编码的声道来重建n声道音频信号。基于核心信号和混合参数（a）进行参数解码，（a）控制空间合成级（150），其被提供有来自下混级（140）的降混信号。选择器（170）以稳态参数和离散解码模式控制解码系统的组件，并在这些模式之间转换。下混合阶段基于n沟道信号（n沟道输入信号或填充有中性值的核心信号）实现对下混合信号的投影。下混合阶段在每个时间帧中是活动的，其中输入信号通过参数编码表示音频信号，并且至少在每个参数编码的时间帧的最后一个时间帧之后的第一时间帧中。

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