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31. 发明申请

US20090136044A1 METHODS AND APPARATUS FOR PROVIDING A DISTINCT PERCEPTUAL LOCATION FOR AN AUDIO SOURCE WITHIN AN AUDIO MIXTURE 有权
标题翻译：用于在音频混合中提供音频源的异常识别位置的方法和装置
公开(公告)号：US20090136044A1
公开(公告)日：2009-05-28
申请号：US11946365
申请日：2007-11-28
申请人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
发明人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
IPC分类号： H04H20/00
CPC分类号： H04S7/30 , H04S7/302 , H04S2400/11
摘要： In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.
摘要翻译：根据用于为音频混合中的音频源提供不同的感知位置的方法，可以处理前景信号以提供前景信号的前景感知角度。还可以处理前景信号以为前景信号提供期望的衰减电平。可以处理背景信号以为背景信号提供背景感知角度。背景信号也可以被处理以为背景信号提供期望的衰减水平。前景信号和背景信号可以组合成输出音频源。

32. 发明授权

US07528745B2 Digital domain sampling rate converter 有权
标题翻译：数字域采样率转换器
公开(公告)号：US07528745B2
公开(公告)日：2009-05-05
申请号：US11452836
申请日：2006-06-13
申请人： Song Wang , Eddie L. T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
发明人： Song Wang , Eddie L. T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
IPC分类号： H03M7/00
CPC分类号： H03H17/0685 , H03H17/0294
摘要： Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter. A sampling rate converter up-samples the digital signal at an input sampling frequency to the selected intermediate sampling frequency, filters the digital signal with the derived anti-aliasing filter, and down-samples the digital signal by the selected down-sampling factor to the desired output sampling frequency.
摘要翻译：描述了通过根据所选择的中间采样频率对数字信号进行上采样和下采样来对数字域中的采样率转换进行描述的技术。具有多个因素的带宽的原型抗混叠滤波器存储在存储器中。这些技术包括基于原型滤波器的因素来选择中间采样频率为数字信号的期望输出采样频率的整数倍，并且将下采样因子选择为与所选择的中间采样相关联的整数频率。滤波器发生器基于原型滤波器生成用于所选择的下采样因子的抗混叠滤波器。采样率转换器将数字信号以输入采样频率向采样频率进行上采样，以采样导出的抗混叠滤波器对数字信号进行滤波，并通过选择的下采样因子将数字信号下采样到所需输出采样频率。

33. 发明申请

US20090089053A1 MULTIPLE MICROPHONE VOICE ACTIVITY DETECTOR 有权
标题翻译：多媒体麦克风语音活动检测器
公开(公告)号：US20090089053A1
公开(公告)日：2009-04-02
申请号：US11864897
申请日：2007-09-28
申请人： Song Wang , Samir Kumar Gupta , Eddie L. T. Choy
发明人： Song Wang , Samir Kumar Gupta , Eddie L. T. Choy
IPC分类号： G10L15/02 , G10L11/02
CPC分类号： G10L25/78 , G10L2021/02165
摘要： Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.
摘要翻译：使用多个麦克风的语音活动检测可以基于语音基准麦克风和噪声参考麦克风各自的能量之间的关系。可以确定来自每个语音参考麦克风和噪声参考麦克风的能量输出。可以确定语音能量比，并将其与预定的语音活动阈值进行比较。在另一个实施例中，确定语音和噪声参考信号的自相关的绝对值，并且确定基于自相关值的比率。超过预定阈值的比率可以指示语音信号的存在。可以使用加权平均值或离散的帧大小来确定语音和噪声能量或自相关性。

34. 发明授权

US07508327B2 Integer representation of relative timing between desired output samples and corresponding input samples 有权
标题翻译：所需输出样本与相应输入样本之间的相对时序的整数表示
公开(公告)号：US07508327B2
公开(公告)日：2009-03-24
申请号：US11558313
申请日：2006-11-09
申请人： Song Wang , Eddie L. T. Choy , Samir Kumar Gupta
发明人： Song Wang , Eddie L. T. Choy , Samir Kumar Gupta
IPC分类号： H03M7/00
CPC分类号： H03H17/0685
摘要： In general, this disclosure describes techniques for changing a sampling frequency of a digital signal. In particular, the techniques provide a more accurate way to determining a relative timing between a desired output sample and a corresponding input sample using a non-approximated integer representation of the relative timing. The relative timing between the desired output sample and corresponding input sample may be represented using a first component that identifies a latest input sample of the digital signal used to generate intermediate samples, a second component that identifies an intermediate sample, and a third component that identifies a timing difference between the desired output sample and the intermediate sample. Each of the components may be recursively updated using non-approximated integer values.
摘要翻译：通常，本公开描述了用于改变数字信号的采样频率的技术。特别地，这些技术提供了使用相对定时的非近似整数表示来确定期望输出采样和相应输入采样之间的相对定时的更精确的方法。可以使用标识用于生成中间样本的数字信号的最新输入样本的第一组件，标识中间样本的第二组件和标识中间样本的第三组件来表示期望输出样本与相应输入样本之间的相对时序所需输出样本和中间样本之间的时间差。可以使用非近似的整数值递归地更新每个组件。

35. 发明授权

US06754630B2 Synthesis of speech from pitch prototype waveforms by time-synchronous waveform interpolation 失效
标题翻译：通过时间 - 同步波形插值从音调原型波形合成语音
公开(公告)号：US06754630B2
公开(公告)日：2004-06-22
申请号：US09191631
申请日：1998-11-13
申请人： Amitava Das , Eddie L. T. Choy
发明人： Amitava Das , Eddie L. T. Choy
IPC分类号： G10L1304
CPC分类号： G10L19/0204 , G10L25/27
摘要： In a method of synthesizing voiced speech from pitch prototype waveforms by time-synchronous waveform interpolation (TSWI), one or more pitch prototypes is extracted from a speech signal or a residue signal. The extraction process is performed in such a way that the prototype has minimum energy at the boundary. Each prototype is circularly shifted so as to be time-synchronous with the original signal. A linear phase shift is applied to each extracted prototype relative to the previously extracted prototype so as to maximize the cross-correlation between successive extracted prototypes. A two-dimensional prototype-evolving surface is constructed by unsampling the prototypes to every sample point. The two-dimensional prototype-evolving surface is re-sampled to generate a one-dimensional, synthesized signal frame with sample points defined by piecewise continuous cubic phase contour functions computed from the pitch lags and the phase shifts added to the extracted prototypes. A pre-selection filter may be applied to determine whether to abandon the TSWI technique in favor of another algorithm for the current frame. A post-selection performance measure may be obtained and compared with a predetermined threshold to determine whether the TSWI algorithm is performing adequately.
摘要翻译：在通过时间 - 同步波形插值（TSWI）从音调原型波形合成有声语音的方法中，从语音信号或残留信号中提取一个或多个音调原型。提取过程以使原型在边界处具有最小能量的方式进行。每个原型都是循环移位的，以便与原始信号保持时间同步。相对于先前提取的原型，对每个提取的原型应用线性相移，以便最大化连续提取的原型之间的互相关。通过对每个采样点的原型进行不抽样来构建二维原型演化曲面。二维原型演化曲面被重新采样以产生一维合成信号帧，其中采样点由从间距延迟计算的分段连续立方相轮廓函数和加到提取的原型上的相移定义。可以应用预选滤波器来确定是否放弃TSWI技术以有利于当前帧的另一算法。可以获得选择后性能测量并与预定阈值进行比较，以确定TSWI算法是否正在充分执行。

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