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21. 发明授权

US09466275B2 Complexity scalable perceptual tempo estimation 有权
标题翻译：复杂性可扩展感知速度估计
公开(公告)号：US09466275B2
公开(公告)日：2016-10-11
申请号：US13503136
申请日：2010-10-26
申请人： Arijit Biswas , Danilo Hollosi , Michael Schug
发明人： Arijit Biswas , Danilo Hollosi , Michael Schug
IPC分类号： G10H7/00 , 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。

22. 发明授权

US09378743B2 Audio encoding method and system for generating a unified bitstream decodable by decoders implementing different decoding protocols 有权
标题翻译：音频编码方法和系统，用于通过实现不同解码协议的解码器生成统一的比特流解码
公开(公告)号：US09378743B2
公开(公告)日：2016-06-28
申请号：US14009503
申请日：2012-04-05
申请人： Jeffrey C. Riedmiller , Farhad Farahani , Michael Schug , Regunathan Radhakrishnan , Mark S. Vinton
发明人： Jeffrey C. Riedmiller , Farhad Farahani , Michael Schug , Regunathan Radhakrishnan , Mark S. Vinton
IPC分类号： G10L19/008 , G10L19/002 , G10L19/16
CPC分类号： G10L19/002 , G10L19/167
摘要： In a class of embodiments, an audio encoding system (typically, a perceptual encoding system that is configured to generate a single (“unified”) bitstream that is compatible with (i.e., decodable by) a first decoder configured to decode audio data encoded in accordance with a first encoding protocol (e.g., the multichannel Dolby Digital Plus, or DD+, protocol) and a second decoder configured to decode audio data encoded in accordance with a second encoding protocol (e.g., the stereo AAC, HE AAC v1, or HE AAC v2 protocol). The unified bitstream can include both encoded data (e.g., bursts of data) decodable by the first decoder (and ignored by the second decoder) and encoded data (e.g., other bursts of data) decodable by the second decoder (and ignored by the first decoder). In effect, the second encoding format is hidden within the unified bitstream when the bitstream is decoded by the first decoder, and the first encoding format is hidden within the unified bitstream when the bitstream is decoded by the second decoder. The format of the unified bitstream generated in accordance with the invention may eliminate the need for transcoding elements throughout an entire media chain and/or ecosystem. Other aspects of the invention are an encoding method performed by any embodiment of the inventive encoder, a decoding method performed by any embodiment of the inventive decoder, and a computer readable medium (e.g., disc) which stores code for implementing any embodiment of the inventive method.
摘要翻译：在一类实施例中，音频编码系统（通常是感知编码系统，其被配置为生成与第一解码器兼容的（即可解码的）单个（“统一”）比特流，第一解码器被配置为对根据第一编码协议（例如，多频道杜比数字+或DD +协议）和被配置为对根据第二编码协议（例如立体声AAC，HE AAC v1或HE）编码的音频数据进行解码的第二解码器统一比特流可以包括可由第一解码器解码（并由第二解码器忽略）的可编码数据（例如，数据突发）和由第二解码器解码的编码数据（例如，其他数据突发）并且被第一解码器忽略），实际上，当第一解码器对比特流进行解码时，第二编码格式被隐藏在统一比特流内，并且当比特流中第一编码格式被隐藏在统一比特流内时令牌由第二解码器解码。根据本发明生成的统一比特流的格式可以消除在整个媒体链和/或生态系统中对代码转换元素的需要。本发明的其他方面是由本发明编码器的任何实施例执行的编码方法，由本发明解码器的任何实施例执行的解码方法，以及存储用于实现本发明的任何实施例的代码的计算机可读介质（例如，盘）方法。

23. 发明申请

US20140358554A1 AUDIO ENCODING METHOD AND SYSTEM FOR GENERATING A UNIFIED BITSTREAM DECODABLE BY DECODERS IMPLEMENTING DIFFERENT DECODING PROTOCOLS 有权
标题翻译：音视频编码方法和系统，用于生成由解码器实现的不同解码协议解码的统一的双绞线
公开(公告)号：US20140358554A1
公开(公告)日：2014-12-04
申请号：US14009503
申请日：2012-04-05
申请人： Jeffrey C. Riedmiller , Farhad Farahani , Michael Schug , Regunathan Radhakrishnan , Mark S. Vinton
发明人： Jeffrey C. Riedmiller , Farhad Farahani , Michael Schug , Regunathan Radhakrishnan , Mark S. Vinton
IPC分类号： G10L19/002
CPC分类号： G10L19/002 , G10L19/167
摘要： In a class of embodiments, an audio encoding system (typically, a perceptual encoding system that is configured to generate a single (“unified”) bitstream that is compatible with (i.e., decodable by) a first decoder configured to decode audio data encoded in accordance with a first encoding protocol (e.g., the multichannel Dolby Digital Plus, or DD+, protocol) and a second decoder configured to decode audio data encoded in accordance with a second encoding protocol (e.g., the stereo AAC, HE AAC v1, or HE AAC v2 protocol). The unified bitstream can include both encoded data (e.g., bursts of data) decodable by the first decoder (and ignored by the second decoder) and encoded data (e.g., other bursts of data) decodable by the second decoder (and ignored by the first decoder). In effect, the second encoding format is hidden within the unified bitstream when the bitstream is decoded by the first decoder, and the first encoding format is hidden within the unified bitstream when the bitstream is decoded by the second decoder. The format of the unified bitstream generated in accordance with the invention may eliminate the need for transcoding elements throughout an entire media chain and/or ecosystem. Other aspects of the invention are an encoding method performed by any embodiment of the inventive encoder, a decoding method performed by any embodiment of the inventive decoder, and a computer readable medium (e.g., disc) which stores code for implementing any embodiment of the inventive method.
摘要翻译：在一类实施例中，音频编码系统（通常是感知编码系统，其被配置为生成与第一解码器兼容的（即可解码的）单个（“统一”）比特流，第一解码器被配置为对根据第一编码协议（例如，多频道杜比数字+或DD +协议）和被配置为对根据第二编码协议（例如立体声AAC，HE AAC v1或HE）编码的音频数据进行解码的第二解码器统一比特流可以包括可由第一解码器解码（并由第二解码器忽略）的可编码数据（例如，数据突发）和由第二解码器解码的编码数据（例如，其他数据突发）并且被第一解码器忽略），实际上，当第一解码器对比特流进行解码时，第二编码格式被隐藏在统一比特流内，并且当比特流中第一编码格式被隐藏在统一比特流内时令牌由第二解码器解码。根据本发明生成的统一比特流的格式可以消除在整个媒体链和/或生态系统中对代码转换元素的需要。本发明的其他方面是由本发明编码器的任何实施例执行的编码方法，由本发明解码器的任何实施例执行的解码方法，以及存储用于实现本发明的任何实施例的代码的计算机可读介质（例如，盘）方法。

24. 发明授权

US08891775B2 Method and encoder for processing a digital stereo audio signal 有权
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：US08891775B2
公开(公告)日：2014-11-18
申请号：US14113362
申请日：2012-05-07
申请人： Michael Schug , Harald H. Mundt
发明人： Michael Schug , Harald H. Mundt
IPC分类号： H04S1/00 , G10L19/008 , G10L19/03
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滤波如果第一和第二预测增益相差超过预定的不匹配范围，则绕过TNS滤波器以获得当前信号帧。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。

25. 发明申请

US20130179175A1 Method and System for Encoding Audio Data with Adaptive Low Frequency Compensation 有权
标题翻译：用自适应低频补偿编码音频数据的方法和系统
公开(公告)号：US20130179175A1
公开(公告)日：2013-07-11
申请号：US13588890
申请日：2012-08-17
申请人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant A. Davidson , Mark S. Vinton
发明人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant A. Davidson , Mark S. Vinton
IPC分类号： G10L19/00
CPC分类号： G10L19/028 , G10L19/0204 , G10L19/032 , G10L19/265
摘要： A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The 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 has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.
摘要翻译：一种用于确定要编码的频域音频数据的尾数位分配的方法，包括通过对数据的一组低频带的每个频带执行自适应低频补偿。低频补偿包括以下步骤：对音频数据执行音调检测，以产生指示该组中的每个频带是否具有突出的音调内容的补偿控制数据; 并且对具有突出音调内容的集合中的每个频带执行低频补偿，包括通过校正具有突出音调内容的每个频带的初步屏蔽值，但不对低频补偿中的任何其他频带中的音频数据执行低频补偿组。其他方面是包括这种音调检测和低频补偿步骤的音频编码方法，以及被配置为执行本发明方法的任何实施例的系统。

26. 发明授权

US08484019B2 Audio encoder and decoder 有权
公开(公告)号：US08484019B2
公开(公告)日：2013-07-09
申请号：US12811421
申请日：2008-12-30
申请人： Per Hedelin , Pontus Carlsson , Jonas Samuelsson , Michael Schug
发明人： Per Hedelin , Pontus Carlsson , Jonas Samuelsson , Michael Schug
IPC分类号： G10L19/00
CPC分类号： G10L19/26 , G10L19/008 , G10L19/032 , G10L19/035
摘要： The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

27. 发明授权

US07318028B2 Method and apparatus for determining an estimate 有权
标题翻译：用于确定估计的方法和装置
公开(公告)号：US07318028B2
公开(公告)日：2008-01-08
申请号：US11469418
申请日：2006-08-31
申请人： Michael Schug , Johannes Hilpert , Stefan Geyersberger , Max Neuendorf
发明人： Michael Schug , Johannes Hilpert , Stefan Geyersberger , Max Neuendorf
IPC分类号： G10L19/00
CPC分类号： G10L19/025 , G10L19/002
摘要： For determining an estimate of a need for information units for encoding a signal, a measure for the distribution of the energy in the frequency band is taken into account in addition to the admissible interference for a frequency band and an energy of the frequency band. With this, a better estimate of the need for information units is obtained, so that coding can be done more efficiently and more accurately.
摘要翻译：为了确定对信号进行编码的信息单元的需求的估计，除了对于频带的容许干扰和频带的能量之外，还考虑了频带中的能量分布的度量。因此，获得对信息单元的更好的估计，使得可以更有效和更准确地进行编码。

28. 发明申请

US20060293884A1 Apparatus and method for determining a quantizer step size 有权
公开(公告)号：US20060293884A1
公开(公告)日：2006-12-28
申请号：US11514006
申请日：2006-08-30
申请人： Bernhard Grill , Michael Schug , Bodo Teichmann , Nikolaus Rettelbach
发明人： Bernhard Grill , Michael Schug , Bodo Teichmann , Nikolaus Rettelbach
IPC分类号： G10L19/00
CPC分类号： G10L19/032 , G10L2019/0005
摘要： For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.

29. 发明申请

US20050175252A1 Device and method for analysing a decoded time signal 有权
标题翻译：用于分析解码时间信号的装置和方法
公开(公告)号：US20050175252A1
公开(公告)日：2005-08-11
申请号：US10220651
申请日：2001-02-16
申请人： Juergen Herre , Martin Dietz , Thomas Sporer , Michael Schug , Wolfgang Schildbach
发明人： Juergen Herre , Martin Dietz , Thomas Sporer , Michael Schug , Wolfgang Schildbach
IPC分类号： H04N19/00 , H04N19/126 , H04N19/40 , H04N19/60 , G06K9/36
CPC分类号： H04N19/00 , H04N19/126 , H04N19/40 , H04N19/60
摘要： An apparatus for analyzing an analysis time signal that has been generated from encoding and decoding an original time signal according to an encoding algorithm first, wherein first the encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. Thereupon, the analysis time signal will be converted from its timely representation comprising a plurality of analysis spectral coefficients, to a spectral representation by using the established encoding block raster. Then, at least two analysis spectral coefficients or at least two spectral coefficients derived from the analysis spectral coefficients by multiplication of an encoding amplification factor or by multiplication with a compression function are grouped. Then, the greatest common divisor of the analysis spectral coefficients or the spectral coefficients derived from the analysis spectral coefficients will be calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. Then, in the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e. for a scale factor band, from the quantization step width. Thus, all parameters used for the quantization of the original time signal are known, so that for quantizing the analysis time signal no longer full iteration loops have to be performed, which are, on the one hand, very computing time intensive and, on the other hand, introduce tandem encoding distortions.
摘要翻译：一种用于分析根据编码算法首先对原始时间信号进行编码和解码而产生的分析时间信号的装置，其中首先确定编码算法使用的分析时间信号下面的编码块光栅。因此，分析时间信号将通过使用所建立的编码块光栅从包括多个分析频谱系数的及时表示转换成频谱表示。然后，将通过编码放大因子的乘法或通过与压缩函数相乘而从分析频谱系数导出的至少两个分析频谱系数或至少两个频谱系数分组。然后，对应于当量化编码算法或其整数倍时使用的量化步长，将计算分析频谱系数的最大公约数或从分析频谱系数导出的频谱系数。然后，在音频信号的情况下，从量化步长可以容易地为该组频谱系数建立比例因子，即缩放因子频带。因此，用于原始时间信号的量化的所有参数是已知的，使得对于分析时间信号的量化不再必须执行完整的迭代循环，这一方面一方面非常计算时间密集，并且在另一方面，引入串联编码失真。

30. 发明授权

US08527264B2 Method and system for encoding audio data with adaptive low frequency compensation 有权
标题翻译：用自适应低频补偿编码音频数据的方法和系统
公开(公告)号：US08527264B2
公开(公告)日：2013-09-03
申请号：US13588890
申请日：2012-08-17
申请人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant Allen Davidson , Mark Stuart Vinton
发明人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant Allen Davidson , Mark Stuart Vinton
IPC分类号： G10L19/00
CPC分类号： G10L19/028 , G10L19/0204 , G10L19/032 , G10L19/265
摘要： A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The 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 has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set; wherein the frequency domain audio data comprises an exponent value for said each low frequency band of the set, and the tonality detection includes determining, for said each low frequency band of the set, a measure of difference between exponents and corresponding tented exponents of the audio data. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.
摘要翻译：一种用于确定要编码的频域音频数据的尾数位分配的方法，包括通过对数据的一组低频带的每个频带执行自适应低频补偿。低频补偿包括以下步骤：对音频数据执行音调检测，以产生指示该组中的每个频带是否具有突出的音调内容的补偿控制数据; 并且对具有突出音调内容的集合中的每个频带执行低频补偿，包括通过校正具有突出音调内容的每个频带的初步屏蔽值，但不对低频补偿中的任何其他频带中的音频数据执行低频补偿组; 其中所述频域音频数据包括所述组的所述每个低频带的指数值，并且所述音调检测包括针对所述组的所述每个低频带确定所述音频的指数和对应的帐篷指数之间的差的度量数据。其他方面是包括这种音调检测和低频补偿步骤的音频编码方法，以及被配置为执行本发明方法的任何实施例的系统。

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