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21. 发明授权

US09378748B2 Reduced complexity converter SNR calculation 有权
标题翻译：降低复杂度转换器SNR计算
公开(公告)号：US09378748B2
公开(公告)日：2016-06-28
申请号：US14439795
申请日：2013-11-04
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING CORPORATION
发明人： Michael Schug , Phillip Williams
IPC分类号： G10L19/00 , G10L19/008 , G10L19/02 , G10L19/032 , G10L19/16
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），其被配置为基于频谱系数集合（312）确定一组缩放因子和一组缩放值（314）; 并编码一组比例因子以产生一组编码比例因子（313）。

22. 发明申请

US20150043754A1 System and Method for Non-Destructively Normalizing Loudness of Audio Signals within Portable Devices 有权
标题翻译：便携式设备中音频信号响度的非破坏性系统和方法
公开(公告)号：US20150043754A1
公开(公告)日：2015-02-12
申请号：US14525536
申请日：2014-10-28
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： Jeffrey Riedmiller , Harald Mundt , Michael Schug , Martin Wolters
IPC分类号： G10L19/02 , H03G3/32
CPC分类号： G10L19/22 , G10L19/02 , G10L19/0208 , G10L19/167 , G10L19/26 , H03G3/3089 , H03G3/32 , H03G7/007
摘要： 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.
摘要翻译：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。播放设备还可以自适应地对播放音频应用增益和限制。在代码转换器和解码器中公开了编码器中的实现。

23. 发明授权

US11670315B2 System and method for non-destructively normalizing loudness of audio signals within portable devices 有权
公开(公告)号：US11670315B2
公开(公告)日：2023-06-06
申请号：US17750803
申请日：2022-05-23
申请人： Dolby Laboratories Licensing Corporation , DOLBY INTERNATIONAL AB
发明人： Jeffrey Riedmiller , Harald Mundt , Michael Schug , Martin Wolters
IPC分类号： G10L19/22 , G10L19/16 , G10L19/02 , H03G7/00 , H03G3/32 , G10L19/26 , H03G3/30
CPC分类号： G10L19/22 , G10L19/02 , G10L19/0208 , G10L19/167 , G10L19/26 , H03G3/3089 , H03G3/32 , H03G7/007
摘要： 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.

24. 发明授权

US10418045B2 System and method for non-destructively normalizing loudness of audio signals within portable devices 有权
公开(公告)号：US10418045B2
公开(公告)日：2019-09-17
申请号：US15482328
申请日：2017-04-07
申请人： DOLBY INTERNATIONAL AB , Dolby Laboratories Licensing Corporation
发明人： Jeffrey Riedmiller , Harald Mundt , Michael Schug , Martin Wolters
IPC分类号： G10L19/22 , G10L19/16 , H03G7/00 , G10L19/02 , H03G3/32 , G10L19/26 , H03G3/30
摘要： 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.

25. 发明申请

US20150269950A1 Reduced Complexity Converter SNR Calculation 有权
公开(公告)号：US20150269950A1
公开(公告)日：2015-09-24
申请号：US14439795
申请日：2013-11-11
申请人： DOLBY INTERNATIONAL AB , DOLBY LABORATORIES LICENSING CORPORATION
发明人： Michael Schug , Phillip Williams
IPC分类号： G10L19/008 , G10L19/032 , G10L19/16 , G10L19/02
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). 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.

26. 发明申请

US20140188488A1 Reduced Complexity Converter SNR Calculation 有权
标题翻译：降低复杂度转换器SNR计算
公开(公告)号：US20140188488A1
公开(公告)日：2014-07-03
申请号：US14184961
申请日：2014-02-20
申请人： Dolby International AB , Dolby Laboratories Licensing Corporation
发明人： Michael Schug , Phillip Williams , Luca Baradel
IPC分类号： G10L19/008
CPC分类号： G10L19/008 , G10L19/02 , G10L19/032 , G10L19/173
摘要： An audio encoder configured to encode an audio signal to generate a bitstream having E-AC-3 format, including by determining a first control parameter indicative of an allocation of available mantissa bits for quantized audio content of the signal. The encoder is configured to perform transcoding simulation to determine a second control parameter in a manner based at least in part on statistical analysis of results of E-AC-3 bit allocation processing of audio data assuming a first target data rate, and of AC-3 bit allocation processing of the data assuming a second target data rate, and to include the second control parameter in the bitstream for use by a converter to convert the bitstream into a second to bitstream having AC-3 format at the second target data rate. Other aspects are converters configured to perform transcoding on a bitstream using such a second control parameter, and methods performed by any embodiment of the inventive encoder or converter.
摘要翻译：音频编码器被配置为对音频信号进行编码以生成具有E-AC-3格式的比特流，包括通过确定指示信号的量化音频内容的可用尾数位的分配的第一控制参数。编码器被配置为执行代码转换模拟，以至少部分地基于假定第一目标数据速率的音频数据的E-AC-3位分配处理的结果的统计分析和AC- 假设第二目标数据速率的数据的3比特分配处理，并且将第二控制参数包括在比特流中以供转换器使用以将比特流转换为具有第二目标数据速率的具有AC-3格式的第二比特流。其他方面是被配置为使用这样的第二控制参数在比特流上执行代码转换的转换器，以及由本发明的编码器或转换器的任何实施例执行的方法。

27. 发明授权

US11232804B2 Low complexity dense transient events detection and coding 有权
公开(公告)号：US11232804B2
公开(公告)日：2022-01-25
申请号：US16628235
申请日：2018-07-03
申请人： Dolby International AB
发明人： Arijit Biswas , Michael Schug , Harald Mundt
IPC分类号： G10L19/025 , G10L19/032 , G10L19/18 , H03M7/30 , H04B1/66 , H04H20/88
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.

28. 发明授权

US09984695B2 Methods and systems for efficient recovery of high frequency audio content 有权
公开(公告)号：US09984695B2
公开(公告)日：2018-05-29
申请号：US15494195
申请日：2017-04-21
申请人： DOLBY INTERNATIONAL AB
发明人： Robin Thesing , Michael Schug
IPC分类号： G10L19/008 , G10L19/16 , G10L19/028
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 for a first frequency subband of an audio signal is described. The first banded tonality value 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 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 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 for the first frequency subband.

29. 发明授权

US09666200B2 Methods and systems for efficient recovery of high frequency audio content 有权
公开(公告)号：US09666200B2
公开(公告)日：2017-05-30
申请号：US14372733
申请日：2013-02-22
申请人： DOLBY INTERNATIONAL AB
发明人： Robin Thesing , Michael Schug
IPC分类号： G10L19/02 , 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.

30. 发明申请

US20140310011A1 Enhanced Chroma Extraction from an Audio Codec 有权
标题翻译：从音频编解码器增强色度提取
公开(公告)号：US20140310011A1
公开(公告)日：2014-10-16
申请号：US14359697
申请日：2012-11-28
申请人： DOLBY INTERNATIONAL AB
发明人： Arijit Biswas , Marco Fink , Michael Schug
IPC分类号： G10L19/02 , G10L19/038 , G10L19/022
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，本文件涉及从音频信号中提取色度矢量的方法和系统。描述了一种用于确定音频信号（301）的采样块的色度矢量（100）的方法（900）。所述方法（900）包括从适于生成的基于频谱带复制的音频编码器（410）的核心编码器（412）接收（901）从所述音频信号（301）的采样块导出的对应的频率系数块根据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用于所述音频信号（301）的采样块的色度矢量（904）（904）。

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