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51. 发明申请

US20100010810A1 POST FILTER AND FILTERING METHOD 审中-公开
标题翻译：后过滤和过滤方法
公开(公告)号：US20100010810A1
公开(公告)日：2010-01-14
申请号：US12518741
申请日：2007-12-13
申请人： Toshiyuki Morii
发明人： Toshiyuki Morii
IPC分类号： G10L11/04
CPC分类号： G10L19/26 , G10L19/125
摘要： When a decoding audio signal is to be acquired by pitch-filtering a combined signal of a sub-frame length, a decoding audio signal is continuously changed at the boundary between sub-frames. The post filter includes: a first filter coefficient calculation unit (306) which obtains a pitch filter coefficient gP(0) of a current frame so as to asymptotically approach the intensity g of the pitch filter from an initial value 0; a second filter coefficient calculation unit (307) which obtains a pitch filter coefficient gP(−1) of a preceding frame so as to asymptotically approach 0 by setting the initial value to the value of the pitch filter coefficient obtained by the first filter coefficient calculation unit (306); a filter state setting unit (308) which sets a pitch filter state fsi for each of the sub-frames; and a pitch filter (309) which pitch-filters the combined signal xi by using the pitch filter coefficients gP(−1), gP(0), and past demodulation audio signals yi−P(−1), yi−P(0).
摘要翻译：当通过对子帧长度的组合信号进行间距滤波来获得解码音频信号时，解码音频信号在子帧之间的边界处连续地改变。后置滤波器包括：第一滤波器系数计算单元（306），其从当前帧获得音调滤波器系数gP（0），以便从初始值0渐近地接近音调滤波器的强度g; 第二滤波器系数计算单元（307），其通过将初始值设置为通过第一滤波器系数计算获得的音调滤波器系数的值来获得前一帧的音调滤波器系数gP（-1），以渐近地接近0 单元（306）; 滤波器状态设置单元（308），其针对每个子帧设置音调滤波器状态fsi; 以及使用音调滤波器系数gP（-1），gP（0）和过去的解调音频信号yi-P（-1），yi-P（0）对组合信号xi进行音调滤波的音调滤波器（309））。

52. 发明授权

US07643988B2 Method for analyzing fundamental frequency information and voice conversion method and system implementing said analysis method 失效
标题翻译：分析基频信息和语音转换方法的方法及系统实现分析方法
公开(公告)号：US07643988B2
公开(公告)日：2010-01-05
申请号：US10551224
申请日：2004-03-02
申请人： Taoufik En-Najjary , Olivier Rosec
发明人： Taoufik En-Najjary , Olivier Rosec
IPC分类号： G10L11/04
CPC分类号： G10L25/90 , G10L25/24 , G10L2021/0135
摘要： A method for analyzing fundamental frequency information contained in voice samples includes at least one analysis step (2) for the voice samples which are grouped together in frames in order to obtain information relating to the spectrum and information relating to the fundamental frequency for each sample frame; a step (20) for the determination of a model representing the common characteristics of the spectrum and fundamental frequency of all samples; and a step (30) for determination of a fundamental frequency prediction function exclusively according to spectrum-related in formation on the basis of the model and voice samples.
摘要翻译：用于分析语音样本中包含的基本频率信息的方法包括至少一个用于语音样本的分析步骤（2），所述语音样本被分组在一起，以获得与频谱有关的信息和与每个样本帧的基频有关的信息 ; 用于确定表示所有样本的频谱和基频的共同特征的模型的步骤（20）; 以及用于根据模型和语音样本专门根据与频谱相关的基频预测函数来确定步骤（30）。

53. 发明申请

US20090204396A1 METHOD AND APPARATUS FOR IMPLEMENTING SPEECH DECODING IN SPEECH DECODER FIELD OF THE INVENTION 有权
标题翻译：用于在语音解码器中实现语音解码的方法和装置技术领域
公开(公告)号：US20090204396A1
公开(公告)日：2009-08-13
申请号：US12426379
申请日：2009-04-20
申请人： Jianfeng Xu , Lijing Xu , Qing Zhang , Wei Li , Shenghu Sang , Zhengzhong Du , Chen Hu
发明人： Jianfeng Xu , Lijing Xu , Qing Zhang , Wei Li , Shenghu Sang , Zhengzhong Du , Chen Hu
IPC分类号： G10L11/04
CPC分类号： G10L19/005 , G10L19/09 , G10L19/107
摘要： The present disclosure relates to a decoding method and apparatus. The method includes: receiving data frames from the coder; if any erroneous frame appears, calculating a pitch lag parameter of the erroneous frame; decoding the data frames according to the calculated pitch lag parameter of the erroneous frame, and obtaining decoded data. The process of determining the pitch lag parameter includes: determining the number of continuous erroneous frames and the pitch lag parameter of the previous frame; adjusting the pitch lag parameter of the previous frame according to the number of the continuous erroneous frames and a preset adjustment policy, and calculating and determining the pitch lag parameter of a current erroneous frame, wherein the preset adjustment policy is adjusting the determined pitch lag parameter of the current erroneous frame within a preset value range according to the number of the continuous erroneous frames.
摘要翻译：本公开涉及一种解码方法和装置。该方法包括：从编码器接收数据帧; 如果出现任何错误帧，则计算错误帧的音调滞后参数; 根据所计算的错误帧的音调滞后参数对数据帧进行解码，并获得解码数据。确定音调滞后参数的过程包括：确定连续错误帧的数量和前一帧的音调滞后参数; 根据连续错误帧的数量和预设的调整策略来调整前一帧的音调滞后参数，以及计算和确定当前错误帧的音调滞后参数，其中预设调整策略正在调整所确定的音调滞后参数根据连续错误帧的数量在预设值范围内的当前错误帧。

54. 发明申请

US20090192788A1 Sound Processing Device and Program 有权
标题翻译：声音处理设备和程序
公开(公告)号：US20090192788A1
公开(公告)日：2009-07-30
申请号：US12358400
申请日：2009-01-23
申请人： Yasuo YOSHIOKA
发明人： Yasuo YOSHIOKA
IPC分类号： G10L11/04 , G10L15/00
CPC分类号： G10L25/78 , G10L25/93
摘要： In a sound processing device, a modulation spectrum specifier specifies a modulation spectrum of an input sound for each of a plurality of unit intervals. An index calculator calculates an index value corresponding to a magnitude of components of modulation frequencies belonging to a predetermined range of the modulation spectrum. A determinator determines whether the input sound of each of the unit intervals is a vocal sound or a non-vocal sound based on the index value. The modulation spectrum specifier analyzes the input sound to obtain a cepstrum or a logarithmic spectrum of the input sound for each of a sequence of frames defined within the unit interval, then specifies a temporal trajectory of a specific component in the cepstrum or the logarithmic spectrum along the sequence of the frames for the unit interval, and performs a Fourier transform on the temporal trajectory throughout the unit interval to thereby specify the modulation spectrum of the unit interval as the result of the Fourier transform of the temporal trajectory.
摘要翻译：在声音处理装置中，调制频谱说明符指定多个单位间隔中的每一个的输入声音的调制频谱。索引计算器计算与属于调制频谱的预定范围的调制频率的分量的幅度相对应的索引值。确定器基于索引值来确定每个单位间隔的输入声音是声音还是非声音。调制频谱指定器分析输入声音以获得在单位间隔内定义的帧序列中的每一帧的输入声音的倒频谱或对数频谱，然后指定倒谱谱中的特定分量或对数频谱中的特征分量的时间轨迹用于单位间隔的帧的序列，并且对整个单位间隔的时间轨迹执行傅里叶变换，从而指定作为时间轨迹的傅立叶变换的结果的单位间隔的调制频谱。

55. 发明申请

US20090182556A1 PITCH ESTIMATION AND MARKING OF A SIGNAL REPRESENTING SPEECH 审中-公开
标题翻译：信号代表声音的估计和标记
公开(公告)号：US20090182556A1
公开(公告)日：2009-07-16
申请号：US12256693
申请日：2008-10-23
申请人： Erik N. Reckase , John F. Remillard
发明人： Erik N. Reckase , John F. Remillard
IPC分类号： G10L11/06 , G10L11/04
CPC分类号： G10L25/93 , G10L25/90
摘要： Methods, systems, and machine-readable media are disclosed for processing a signal representing speech. According to one embodiment, a method of processing a signal representing speech can comprise receiving a frame of the signal representing speech, classifying the frame as a voiced frame, and parsing the voiced frame into one or more regions based on occurrence of one or more events within the voiced frame. For example, the one or more events can comprise one or more glottal pulses. The one or more regions may collectively represent less than all of the voiced frame.
摘要翻译：公开了用于处理表示语音的信号的方法，系统和机器可读介质。根据一个实施例，一种处理表示语音的信号的方法可以包括接收表示语音的信号的帧，将帧分类为有声帧，以及基于一个或多个事件的发生将有声帧解析成一个或多个区域在声音框架内。例如，一个或多个事件可以包括一个或多个声门脉冲。一个或多个区域可以共同地表示小于所有有声帧的全部。

56. 发明授权

US07552048B2 Method and device for performing frame erasure concealment on higher-band signal 有权
标题翻译：在较高频带信号上执行帧擦除隐藏的方法和装置
公开(公告)号：US07552048B2
公开(公告)日：2009-06-23
申请号：US12273391
申请日：2008-11-18
申请人： Jianfeng Xu , Lei Miao , Chen Hu , Qing Zhang , Lijing Xu , Wei Li , Zhengzhong Du , Yi Yang , Fengyan Qi , Wuzhou Zhan , Dongqi Wang
发明人： Jianfeng Xu , Lei Miao , Chen Hu , Qing Zhang , Lijing Xu , Wei Li , Zhengzhong Du , Yi Yang , Fengyan Qi , Wuzhou Zhan , Dongqi Wang
IPC分类号： G10L11/04 , G10L19/00
CPC分类号： G10L19/005 , G10L19/0204
摘要： A method for performing a frame erasure concealment for a higher-band signal involves calculating a periodic intensity of the higher-band signal with respect to pitch period information of a lower-band signal; comparing the periodic intensity to a preconfigured threshold and, if the periodic intensity is greater or equal to the preconfigured threshold, performing the frame erasure concealment with a pitch period repetition based method. If the periodic intensity is less than the preconfigured threshold, performing the frame erasure concealment with a previous frame data repetition based method. A device for performing a frame erasure concealment includes a periodic intensity calculation module, a pitch period repetition module, and a previous frame data repetition module. The pitch period repetition module performs the frame erasure concealment with a pitch period repetition based method; and the previous frame data repetition module performs the frame erasure concealment with a previous frame data repetition based method.
摘要翻译：用于执行较高频带信号的帧擦除隐藏的方法包括：计算相对于较低频带信号的音调周期信息的较高频带信号的周期性强度; 将周期性强度与预配置的阈值进行比较，并且如果周期性强度大于或等于预配置阈值，则以基于音调周期重复的方法执行帧擦除隐藏。如果周期性强度小于预配置阈值，则使用先前基于帧数据重复的方法来执行帧擦除隐藏。用于执行帧擦除隐藏的装置包括周期性强度计算模块，音调周期重复模块和先前的帧数据重复模块。音调周期重复模块以音调周期重复方式执行帧擦除隐藏; 前一帧数据重复模块利用先前基于帧数据重复的方法执行帧擦除隐藏。

57. 发明申请

US20090144053A1 SPEECH PROCESSING APPARATUS AND SPEECH SYNTHESIS APPARATUS 有权
标题翻译：语音处理设备和语音合成设备
公开(公告)号：US20090144053A1
公开(公告)日：2009-06-04
申请号：US12327399
申请日：2008-12-03
申请人： Masatsune TAMURA , Katsumi TSUCHIYA , Takehiko KAGOSHIMA
发明人： Masatsune TAMURA , Katsumi TSUCHIYA , Takehiko KAGOSHIMA
IPC分类号： G10L13/00 , G10L11/04 , G10L13/08
CPC分类号： G10L13/06
摘要： An information extraction unit extracts spectral envelope information of L-dimension from each frame of speech data. The spectral envelope information does not have a spectral fine structure. A basis storage unit stores N bases (L>N>1). Each basis is differently a frequency band having a maximum as a peak frequency in a spectral domain having L-dimension. A value corresponding to a frequency outside the frequency band along a frequency axis of the spectral domain is zero. Two frequency bands of which two peak frequencies are adjacent along the frequency axis partially overlap. A parameter calculation unit minimizes a distortion between the spectral envelope information and a linear combination of each basis with a coefficient by changing the coefficient, and sets the coefficient of each basis from which the distortion is minimized to a spectral envelope parameter of the spectral envelope information.
摘要翻译：信息提取单元从每个语音数据帧提取L维的频谱包络信息。光谱包络信息不具有光谱精细结构。基准存储单元存储N个碱基（L> N> 1）。每个基准在具有L维的谱域中具有作为峰值频率的最大值的频带不同。对应于沿着频域的频率轴的频带外的频率的值为零。两个峰值频率沿频率轴相邻的两个频带部分重叠。参数计算单元通过改变系数，将频谱包络信息和每个基线的线性组合之间的失真最小化为系数，并且将将失真最小化的每个基础的系数设置为频谱包络信息的频谱包络参数。

58. 发明授权

US07533015B2 Signal enhancement via noise reduction for speech recognition 失效
标题翻译：通过语音识别降噪的信号增强
公开(公告)号：US07533015B2
公开(公告)日：2009-05-12
申请号：US11067809
申请日：2005-02-28
申请人： Tetsuya Takiguchi , Masafumi Nishimura
发明人： Tetsuya Takiguchi , Masafumi Nishimura
IPC分类号： G10L11/04
CPC分类号： G10L21/0208
摘要： Provides speech enhancement techniques for extemporaneous noise without a noise interval and unknown extemporaneous noise. Signal enhancement includes: subtracting a given reference signal from an input signal containing a target signal and a noise signal by spectral subtraction; applying an adaptive filter to the reference signal; and controlling a filter coefficient of the adaptive filter in order to reduce components of the noise signal in the input signal. In signal enhancement, a database of a signal model concerning the target signal expressing a given feature by a given statistical model is provided, and the filter coefficient is controlled based on the likelihood of the signal model with respect to an output signal from the spectral subtraction means.
摘要翻译：提供语音增强技术，用于即时噪声，无噪声间隔和未知的即时噪声。信号增强包括：通过频谱减法从包含目标信号的输入信号和噪声信号中减去给定的参考信号; 对参考信号应用自适应滤波器; 以及控制自适应滤波器的滤波器系数，以便减少输入信号中噪声信号的分量。在信号增强中，提供了关于通过给定统计模型表示给定特征的目标信号的信号模型的数据库，并且基于信号模型相对于来自频谱相减的输出信号的似然性来控制滤波器系数手段。

59. 发明申请

US20090119096A1 PARTIAL SPEECH RECONSTRUCTION 有权
标题翻译：部分语音重建
公开(公告)号：US20090119096A1
公开(公告)日：2009-05-07
申请号：US12254488
申请日：2008-10-20
申请人： Franz Gerl , Tobias Herbig , Mohamed Krini , Gerhard Uwe Schmidt
发明人： Franz Gerl , Tobias Herbig , Mohamed Krini , Gerhard Uwe Schmidt
IPC分类号： G10L11/04 , G10L21/02 , G06F17/00
CPC分类号： H04R3/005 , G10L21/0208 , G10L21/0264 , G10L2021/02165 , H04R2410/05 , H04R2410/07 , H04R2499/11 , H04R2499/13
摘要： A system enhances the quality of a digital speech signal that may include noise. The system identifies vocal expressions that correspond to the digital speech signal. A signal-to-noise ratio of the digital speech signal is measured before a portion of the digital speech signal is synthesized. The selected portion of the digital speech signal may have a signal-to-noise ratio below a predetermined level and the synthesis of the digital speech signal may be based on speaker identification.
摘要翻译：系统提高可能包括噪声的数字语音信号的质量。该系统识别对应于数字语音信号的声乐表达。在数字语音信号的一部分被合成之前测量数字语音信号的信噪比。数字语音信号的所选部分可以具有低于预定电平的信噪比，并且数字语音信号的合成可以基于说话者识别。

60. 发明授权

US07493254B2 Pitch determination method and apparatus using spectral analysis 失效
标题翻译：使用频谱分析的音调确定方法和装置
公开(公告)号：US07493254B2
公开(公告)日：2009-02-17
申请号：US10486065
申请日：2002-08-08
申请人： Doill Jung , Hunseok Seo
发明人： Doill Jung , Hunseok Seo
IPC分类号： G10L11/04 , G10G7/02
CPC分类号： G10L25/90
摘要： A method and apparatus for detecting a pitch using frequency analysis are provided. An externally input digital signal is analyzed into frequency component values at predetermined time intervals, and positions of peaks of the digital signal are detected based on the frequency component values. It is determined whether a frequency at a maximum peak position among the peak positions is a pitch or a n-order harmonic frequency of the pitch to detect a pitch. Then, the range of the pitch is determined based on the range of a harmonic frequency of the detected pitch. Accordingly, an error range for the pitch detected using frequency analysis is minimized, thereby more exactly detecting a pitch when the pitch is detected using the frequency analysis.
摘要翻译：提供了使用频率分析来检测音调的方法和装置。外部输入的数字信号以预定的时间间隔被分析成频率分量值，并且基于频率分量值来检测数字信号的峰值的位置。确定峰值位置之间的最大峰值位置处的频率是否是用于检测音高的间距或n次谐波频率。然后，基于检测到的音调的谐波频率的范围来确定音调的范围。因此，使用频率分析检测的音调的误差范围被最小化，从而更精确地检测使用频率分析检测音调时的音调。

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