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1. 发明授权

US09135929B2 Efficient content classification and loudness estimation 有权
标题翻译：高效的内容分类和响度估计
公开(公告)号：US09135929B2
公开(公告)日：2015-09-15
申请号：US14112537
申请日：2012-04-27
申请人： Harald H. Mundt , Arijit Biswas , Rolf Meissner
发明人： Harald H. Mundt , Arijit Biswas , Rolf Meissner
IPC分类号： G10L19/00 , G10L25/93 , G10L19/16 , G10L19/24 , G10L25/78
CPC分类号： G10L25/93 , G10L19/167 , G10L19/24 , G10L2025/783
摘要： Efficient Context Classification and Gated Loudness Estimation The present document relates to methods and systems for encoding an audio signal. The method comprises determining a spectral representation of the audio signal. The determining a spectral representation step may comprise determining modified discrete cosine transform, MDCT, coefficients, or a Quadrature Mirror Filter, QMF, filter bank representation of the audio signal. The method further comprises encoding the audio signal using the determined spectral representation; and classifying parts of the audio signal to be speech or non-speech based on the determined spectral representation. Finally, a loudness measure for the audio signal based on the speech parts is determined.
摘要翻译：高效上下文分类和门控响度估计本文件涉及用于对音频信号进行编码的方法和系统。该方法包括确定音频信号的频谱表示。确定频谱表示步骤可以包括确定音频信号的修正离散余弦变换，MDCT，系数或正交镜像滤波器QMF滤波器组表示。该方法还包括使用所确定的频谱表示对音频信号进行编码; 以及基于所确定的频谱表示将音频信号的部分分类为语音或非语音。最后，确定基于语音部分的音频信号的响度测量。

2. 发明授权

US09117440B2 Method, apparatus, and medium for detecting frequency extension coding in the coding history of an audio signal 有权
标题翻译：用于在音频信号的编码历史中检测频率扩展编码的方法，装置和介质
公开(公告)号：US09117440B2
公开(公告)日：2015-08-25
申请号：US14116113
申请日：2012-04-30
申请人： Harald H. Mundt , Arijit Biswas , Regunathan Radhakrishnan
发明人： Harald H. Mundt , Arijit Biswas , Regunathan Radhakrishnan
IPC分类号： G10L19/00 , G10L21/02 , G10L25/03 , G10L19/008 , G10L21/038
CPC分类号： G10L19/00 , G10L19/008 , G10L21/02 , G10L21/038 , G10L25/03
摘要： The present document relates to audio forensics, notably the blind detection of traces of parametric audio encoding/decoding. In particular, the present document relates to the detection of parametric frequency extension audio coding, such as spectral band replication (SBR) or spectral extension (SPX), from uncompressed waveforms such as PCM (pulse code modulation) encoded waveforms. A method for detecting frequency extension coding history in a time domain audio signal is described. The method may comprise transforming the time domain audio signal into a frequency domain, thereby generating a plurality of subband signals in a corresponding plurality of subbands comprising low and high frequency subbands; determining a degree of relationship between subband signals in the low frequency subbands and subband signals in the high frequency subbands; wherein the degree of relationship is determined based on the plurality of subband signals; and determining frequency extension coding history if the degree of relationship is greater than a relationship threshold.
摘要翻译：本文件涉及音频取证，特别是盲目检测参数音频编码/解码的痕迹。特别地，本文件涉及从诸如PCM（脉冲编码调制）编码波形的未压缩波形检测参数频率扩展音频编码，例如频谱带复制（SBR）或频谱扩展（SPX）。描述了用于检测时域音频信号中的频率扩展编码历史的方法。该方法可以包括将时域音频信号变换成频域，从而在包括低频和高频子带的相应多个子带中产生多个子带信号; 确定低频子带中的子带信号与高频子带中的子带信号之间的关系程度; 其中所述关系度基于所述多个子带信号来确定; 以及如果所述关系度大于关系阈值，则确定频率扩展编码历史。

3. 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。

4. 发明申请

US20140072120A1 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：US20140072120A1
公开(公告)日：2014-03-13
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
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。

5. 发明申请

US20140039890A1 EFFICIENT CONTENT CLASSIFICATION AND LOUDNESS ESTIMATION 有权
标题翻译：有效内容分类和LOUDNESS估计
公开(公告)号：US20140039890A1
公开(公告)日：2014-02-06
申请号：US14112537
申请日：2012-04-27
申请人： Harald H. Mundt , Arijit Biswas , Rolf Meissner
发明人： Harald H. Mundt , Arijit Biswas , Rolf Meissner
IPC分类号： G10L25/93
CPC分类号： G10L25/93 , G10L19/167 , G10L19/24 , G10L2025/783
摘要： The present document relates to methods and systems for encoding an audio signal. The method comprises determining a spectral representation of the audio signal. The determining a spectral representation step may comprise determining modified discrete cosine transform, MDCT, coefficients, or a Quadrature Mirror Filter, QMF, filter bank representation of the audio signal. The method further comprises encoding the audio signal using the determined spectral representation; and classifying parts of the audio signal to be speech or non-speech based on the determined spectral representation. Finally, a loudness measure for the audio signal based on the speech parts is determined.
摘要翻译：本文件涉及用于对音频信号进行编码的方法和系统。该方法包括确定音频信号的频谱表示。确定频谱表示步骤可以包括确定音频信号的修正离散余弦变换，MDCT，系数或正交镜像滤波器QMF滤波器组表示。该方法还包括使用所确定的频谱表示对音频信号进行编码; 以及基于所确定的频谱表示将音频信号的部分分类为语音或非语音。最后，确定基于语音部分的音频信号的响度测量。

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