公开(公告)号：WO2014072260A2

公开(公告)日：2014-05-15

申请号：PCT/EP2013/072961

申请日：2013-11-04

Applicant: DOLBY INTERNATIONAL AB ,  DOLBY LABORATORIES LICENSING CORPORATION

Inventor： SCHUG, Michael ,  WILLIAMS, Phillip

IPC: G10L19/03

CPC classification number: G10L19/008 ,  G10L19/02 ,  G10L19/032 ,  G10L19/173

Abstract: 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). In addition, the encoder (300) comprises a bit allocation and quantization unit (305, 306) configured to determine a total number of available bits for quantizing the set of scaled values (314), based on a first target data-rate and based on the number of bits used for the set of encoded scale factors (313); to determine a first control parameter (315) indicative of an allocation of the total number of available bits for quantizing the scaled values of the set of scaled values (314); and to quantize the set of scaled values (314) in accordance to the first control parameter (315) to yield a set of quantized scaled values (317). Furthermore, the encoder (300) comprises a transcoding simulation unit (320) configured to determine a second control parameter (321) based on the first control parameter (315); wherein the second control parameter (321) enables a transcoder to convert the first bitstream into a second bitstream at a second target data-rate; wherein the second bitstream accords to a second audio codec system different from the first audio codec system; and wherein the first bitstream comprises the second control parameter.

Abstract translation: 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处理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），该浮点编码单元（304）被配置为基于该组频谱系数（312）确定一组缩放因子和一组缩放值（314）。 并对该组比例因子进行编码以产生一组编码比例因子（313）。 另外，编码器（300）包括比特分配和量化单元（305,306），该比特分配和量化单元被配置为基于第一目标数据速率来确定用于量化该组缩放值（314）的可用比特的总数 用于该组编码比例因子的比特数（313）; 确定指示可用比特总数的分配的第一控制参数（315），用于量化该组缩放值（314）的缩放值; 并且根据第一控制参数（315）量化该组缩放值（314）以产生一组量化缩放值（317）。 此外，编码器（300）包括被配置为基于第一控制参数（315）确定第二控制参数（321）的转码模拟单元（320）; 其中所述第二控制参数（321）使代码转换器能够以第二目标数据速率将所述第一比特流转换为第二比特流; 其中所述第二比特流符合不同于所述第一音频编解码器系统的第二音频编解码器系统; 并且其中第一比特流包括第二控制参数。

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

Applicant: DOLBY INTERNATIONAL AB

Inventor： THESING, Robin ,  SCHUG, Michael

IPC: G10L21/0388

CPC classification number: G10L19/008 ,  G10L19/0204 ,  G10L19/028 ,  G10L19/167 ,  G10L21/0388

Abstract: 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.

Abstract translation: 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。

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

Applicant: DOLBY INTERNATIONAL AB ,  BOEHM, Reinhold ,  GROESCHEL, Alexander ,  HOERICH, Holger ,  HOMM, Daniel ,  SCHILDBACH, Wolfgang A. ,  SCHUG, Michael ,  WATZKE, Oliver ,  WOLTERS, Martin ,  ZIEGLER, Thomas

Inventor： BOEHM, Reinhold ,  GROESCHEL, Alexander ,  HOERICH, Holger ,  HOMM, Daniel ,  SCHILDBACH, Wolfgang A. ,  SCHUG, Michael ,  WATZKE, Oliver ,  WOLTERS, Martin ,  ZIEGLER, Thomas

IPC: G11B20/00 ,  G06F21/00 ,  H04L29/06 ,  H04N7/167 ,  H04N7/24 ,  H04N7/52 ,  H04H60/56

CPC classification number: 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

Abstract: 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.

Abstract translation: 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。

公开(公告)号：WO2013106098A1

公开(公告)日：2013-07-18

申请号：PCT/US2012/057132

申请日：2012-09-25

Applicant: DOLBY LABORATORIES LICENSING CORPORATION ,  DOLBY INTERNATIONAL AB

Inventor： BISWAS, Arijit ,  MELKOTE, Vinay ,  SCHUG, Michael ,  DAVIDSON, Grant A. ,  VINTON, Mark S.

IPC: G10L19/032 ,  G10L19/02

CPC classification number: G10L19/028 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/265

Abstract: A method for determining mantissa bit allocation of audio data values of frequency domain audio data to be encoded. The allocation method includes a step of determining masking values for the audio data values, including by performing adaptive low frequency compensation on the audio data of each frequency band of a set of low frequency bands of the audio data. The adaptive low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set of low frequency bands has prominent tonal content; and performing low frequency compensation on the audio data in each frequency band in the set of low frequency bands having prominent tonal content as indicated by the compensation control data, but not performing low frequency compensation on the audio data in any other frequency band in the set of low frequency bands.

Abstract translation: 一种用于确定要编码的频域音频数据的音频数据值的尾数位分配的方法。 分配方法包括通过对音频数据的一组低频带的每个频带的音频数据执行自适应低频补偿来确定音频数据值的屏蔽值的步骤。 所述自适应低频补偿包括以下步骤：对所述音频数据执行音调检测，以产生指示所述一组低频带中的每个频带是否具有突出的音调内容的补偿控制数据; 对由该补偿控制数据所表示的具有突出色调内容的低频带组中的每个频带中的音频数据执行低频补偿，而不对该组中的任何其它频带中的音频数据执行低频补偿 的低频带。

公开(公告)号：WO2011110525A1

公开(公告)日：2011-09-15

申请号：PCT/EP2011/053389

申请日：2011-03-07

Applicant: DOLBY INTERNATIONAL AB ,  DOLBY LABORATORIES LICENSING CORPORATION ,  WOLTERS, Martin ,  SCHUG, Michael ,  MUNDT, Harald, H. ,  RIEDMILLER, Jeffrey, C.

Inventor： WOLTERS, Martin ,  SCHUG, Michael ,  MUNDT, Harald, H. ,  RIEDMILLER, Jeffrey, C.

IPC: H03G9/00 ,  H03G9/14

CPC classification number: H03G9/00 ,  H03G9/005 ,  H03G9/14

Abstract: The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

Abstract translation: 本文件涉及多媒体数据的处理，特别是多媒体数据的编码，传输，解码和呈现，例如， 音频文件或比特流。 特别地，本文件涉及多媒体播放器中的响度控制的实现。 描述了一种用于向媒体播放器提供响度相关数据的方法。 该方法包括以下步骤：提供与音频信号相关联的第一响度相关值; 其中所述第一响度相关值已经根据第一过程确定; 使用可逆关系将第一响度相关值转换为第二响度相关值; 其中所述第二响度相关值与用于确定响度相关值的第二过程相关联; 将第二响度相关值存储在与音频信号相关联的元数据中; 并向媒体播放器提供元数据。

公开(公告)号：WO2011100155A1

公开(公告)日：2011-08-18

申请号：PCT/US2011/023531

申请日：2011-02-03

Applicant: DOLBY LABORATORIES LICENSING CORPORATION ,  DOLBY INTERNATIONAL AB ,  RIEDMILLER, Jeffrey C. ,  MUNDT, Harald H. ,  SCHUG, Michael ,  WOLTERS, Martin

Inventor： RIEDMILLER, Jeffrey C. ,  MUNDT, Harald H. ,  SCHUG, Michael ,  WOLTERS, Martin

IPC: G10L19/14 ,  H03G7/00

CPC classification number: G10L19/22 ,  G10L19/02 ,  G10L19/0208 ,  G10L19/167 ,  G10L19/26 ,  H03G3/3089 ,  H03G3/32 ,  H03G7/007

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract translation: 许多便携式播放设备不能解码和播放具有宽带宽和宽动态范围的编码音频内容，具有一致的响度和清晰度，除非编码音频内容已经为这些设备专门准备。 通过使用编码的内容包含一些通过相对于另一已知压缩简档的绝对值或差分值来指定合适的动态范围压缩简档的元数据来克服该问题。 播放装置还可以自适应地对播放音频应用增益和限制。 公开了在编码器，代码转换器和解码器中的实现。

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

Applicant: DOLBY INTERNATIONAL AB

Inventor： BISWAS, Arijit ,  SCHUG, Michael ,  MUNDT, Harald

IPC: G10L19/025 ,  G10L19/03

CPC classification number: G10L19/025 ,  G10L19/03

Abstract: 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.

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

Applicant: DOLBY INTERNATIONAL AB

Inventor： BISWAS, Arijit ,  FINK, Marco ,  SCHUG, Michael

IPC: G10L25/48

CPC classification number: G10L19/02 ,  G10H1/0008 ,  G10H1/383 ,  G10H2210/066 ,  G10H2250/225 ,  G10L19/022 ,  G10L19/038 ,  G10L21/0388 ,  G10L25/54

Abstract: 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.

Abstract translation: 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）。

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

Applicant: DOLBY INTERNATIONAL AB ,  SCHUG, Michael ,  MUNDT, Harald

Inventor： SCHUG, Michael ,  MUNDT, Harald

IPC: G10L19/00 ,  G10L19/02

CPC classification number: H04S1/007 ,  G10L19/008 ,  G10L19/03

Abstract: 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.

Abstract translation: 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。

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

Applicant: DOLBY INTERNATIONAL AB ,  BISWAS, Arijit ,  HOLLOSI, Danilo ,  SCHUG, Michael

Inventor： BISWAS, Arijit ,  HOLLOSI, Danilo ,  SCHUG, Michael

IPC: G10H1/40

CPC classification number: G10H1/40 ,  G10H2210/076 ,  G10H2230/015 ,  G10H2240/075

Abstract: 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.

Abstract translation: 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。