公开(公告)号：US20080201150A1

公开(公告)日：2008-08-21

申请号：US12017740

申请日：2008-01-22

Applicant: Masatsune Tamura ,  Takehiro Kagoshima

Inventor： Masatsune Tamura ,  Takehiro Kagoshima

IPC: G10L13/06

CPC classification number: G10L21/00 ,  G10L2021/0135

Abstract: A conversion rule and a rule selection parameter are stored. The conversion rule converts a spectral parameter of a source speaker to a spectral parameter of a target speaker. The rule selection parameter represents the spectral parameter of the source speaker. A first conversion rule of start timing and a second conversion rule of end timing in a speech unit of the source speaker are selected by the spectral parameter of the start timing and the end timing. An interpolation coefficient corresponding to the spectral parameter of each timing in the speech unit is calculated by the first conversion rule and the second conversion rule. A third conversion rule corresponding to the spectral parameter of each timing in the speech unit is calculated by interpolating the first conversion rule and the second conversion rule with the interpolation coefficient. The spectral parameter of each timing is converted to a spectral parameter of the target speaker by the third conversion rule. A spectral acquired from the spectral parameter of the target speaker is compensated by a spectral compensation quantity. A speech waveform is generated from the compensated spectral.

Abstract translation: 存储转换规则和规则选择参数。 转换规则将源扬声器的频谱参数转换为目标扬声器的频谱参数。 规则选择参数表示源扬声器的频谱参数。 通过开始定时和结束定时的频谱参数来选择源扬声器的语音单元中的开始定时和结束定时的第二转换规则的第一转换规则。 通过第一转换规则和第二转换规则来计算对应于语音单元中每个定时的频谱参数的内插系数。 通过用内插系数内插第一转换规则和第二转换规则来计算对应于语音单元中每个定时的频谱参数的第三转换规则。 通过第三转换规则将每个定时的频谱参数转换成目标说话者的频谱参数。 从目标扬声器的频谱参数获得的频谱由频谱补偿量补偿。 从补偿光谱生成语音波形。

公开(公告)号：US20080147385A1

公开(公告)日：2008-06-19

申请号：US11611798

申请日：2006-12-15

Applicant: Jani Nurminen ,  Jilei Tian ,  Victor Popa

Inventor： Jani Nurminen ,  Jilei Tian ,  Victor Popa

IPC: G10L19/12

CPC classification number: G10L21/00 ,  G10L2021/0135

Abstract: An improved system method for enabling and implementing codebook-based voice conversion that both significantly reduces the memory footprint and improves the continuity of the output. In various embodiments, the paired source-target codebook is implemented as a multi-stage vector quantizer. During the conversion, N best candidates in a tree search are taken as the output from the quantizer. The N candidates for each vector to be converted are used in a dynamic programming-based approach that finds a smooth but accurate output sequence.

Abstract translation: 一种改进的系统方法，用于启用和实施基于代码本的语音转换，可显着减少内存占用并提高输出的连续性。 在各种实施例中，成对的源目标码本被实现为多级矢量量化器。 在转换期间，树搜索中的N个最佳候选者作为量化器的输出。 将要转换的每个向量的N个候选者用于基于动态规划的方法，其寻找平滑但准确的输出序列。

公开(公告)号：US20080120115A1

公开(公告)日：2008-05-22

申请号：US11600938

申请日：2006-11-16

Applicant: Xiao Dong Mao

Inventor： Xiao Dong Mao

IPC: G10L21/00

CPC classification number: G10L21/00 ,  G10L2021/0135

Abstract: In one embodiment, the methods and apparatuses detect an original audio signal;detect a sound model wherein the sound model includes a sound parameter; transform the original audio signal based on the parameter whereby forming a transformed audio signal; and compare the transformed audio signal with the original audio signal.

Abstract translation: 在一个实施例中，方法和装置检测原始音频信号;检测声音模型，其中声音模型包括声音参数; 基于参数来变换原始音频信号，从而形成变换后的音频信号; 并将变换的音频信号与原始音频信号进行比较。

公开(公告)号：US20070239634A1

公开(公告)日：2007-10-11

申请号：US11400629

申请日：2006-04-07

Applicant: Jilei Tian ,  Jani Nurminen ,  Victor Popa

Inventor： Jilei Tian ,  Jani Nurminen ,  Victor Popa

IPC: G06N3/02

CPC classification number: G10L21/00 ,  G10L13/033 ,  G10L2021/0135

Abstract: An apparatus for providing efficient evaluation of feature transformation includes a training module and a transformation module. The training module is configured to train a Gaussian mixture model (GMM) using training source data and training target data. The transformation module is in communication with the training module. The transformation module is configured to produce a conversion function in response to the training of the GMM. The training module is further configured to determine a quality of the conversion function prior to use of the conversion function by calculating a trace measurement of the GMM.

Abstract translation: 用于提供特征变换的有效评估的装置包括训练模块和变换模块。 训练模块被配置为使用训练源数据和训练目标数据训练高斯混合模型（GMM）。 变换模块与训练模块通信。 转换模块被配置为响应于GMM的训练而产生转换功能。 训练模块还被配置为通过计算GMM的跟踪测量来确定在使用转换功能之前的转换功能的质量。

公开(公告)号：US20070192100A1

公开(公告)日：2007-08-16

申请号：US10591599

申请日：2005-03-14

Applicant: Olivier Rosec ,  Taoufik En-Najjary

Inventor： Olivier Rosec ,  Taoufik En-Najjary

IPC: G10L17/00

CPC classification number: G10L21/00 ,  G10L2021/0135

Abstract: A method for converting a voice signal from a source speaker into a converted voice signal with acoustic characteristics similar to those of a target speaker includes the steps of determining (1) at least one function for transforming source speaker acoustic characteristics into acoustic characteristics similar to those of the target speaker using target and source speaker voice samples; and transforming acoustic characteristics of the source speaker voice signal to be converted by applying the transformation function(s). The method is characterized in that the transformation (2) includes the step (44) of applying only a predetermined portion of at least one transformation function to said signal to be converted.

Abstract translation: 用于将来自扬声器的语音信号转换为具有与目标扬声器相似的声学特性的转换语音信号的方法包括以下步骤：确定（1）至少一个功能，用于将源扬声器声学特性变换成类似于声音特性的声学特性 使用目标和源扬声器语音样本的目标扬声器; 以及通过应用变换函数来转换要转换的源扬声器语音信号的声学特性。 该方法的特征在于，变换（2）包括仅将至少一个变换函数的预定部分应用于所述要转换的信号的步骤（44）。

公开(公告)号：US07191127B2

公开(公告)日：2007-03-13

申请号：US10328687

申请日：2002-12-23

Applicant: Geydi Lorenzo ,  Charles D. Estes

Inventor： Geydi Lorenzo ,  Charles D. Estes

IPC: G10L15/20

CPC classification number: H04M1/6025 ,  G10L21/0364 ,  G10L2021/0135 ,  H04R25/00 ,  H04R2225/43

Abstract: A method and apparatus for reducing noise in a speech signal. A handset or remote unit provides to users with a hearing deficiency, a first mode of operation where noise suppressant/speech enhancement algorithms are used during any auditory-related service. There is also provided, in a related mode of operation, speech filtering for reducing noise in a speech signal received through the microphone and outputting the filtered sound to the speaker. The handset includes a microphone for receiving an auditory sound, a receiver for receiving an auditory signal and a speech filter for suppressing noise in the auditory signal and sound. The speech filter also may be configured to shift the frequency and/or alter the intensity of the auditory signal and sound. The speaker is used for amplifying and outputting the enhanced speech component as an audible sound.

Abstract translation: 一种降低语音信号噪声的方法和装置。 手机或远程单元向用户提供听力不足，在任何听觉相关服务中使用噪声抑制/语音增强算法的第一操作模式。 在相关操作模式中还提供语音滤波，用于降低通过麦克风接收的语音信号中的噪声并将滤波的声音输出到扬声器。 手机包括用于接收听觉声音的麦克风，用于接收听觉信号的接收器和用于抑制听觉信号和声音中的噪声的语音滤波器。 语音滤波器还可以被配置为移动频率和/或改变听觉信号和声音的强度。 扬声器用于放大并输出增强语音分量作为可听见的声音。

公开(公告)号：US20060178874A1

公开(公告)日：2006-08-10

申请号：US10551224

申请日：2004-03-02

Applicant: Taoufik En-Najjary ,  Olivier Rosec

Inventor： Taoufik En-Najjary ,  Olivier Rosec

IPC: G10L11/04

CPC classification number: G10L25/90 ,  G10L25/24 ,  G10L2021/0135

Abstract: 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.

Abstract translation: 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）。

公开(公告)号：US07085719B1

公开(公告)日：2006-08-01

申请号：US09615266

申请日：2000-07-13

Applicant: Craig R. Shambaugh ,  Anthony Dezonno ,  Mark J. Power ,  Kenneth Venner ,  James Martin ,  Darryl Hymel ,  Laird C. Williams

Inventor： Craig R. Shambaugh ,  Anthony Dezonno ,  Mark J. Power ,  Kenneth Venner ,  James Martin ,  Darryl Hymel ,  Laird C. Williams

IPC: G01L15/04 ,  H04M3/22

CPC classification number: G10L13/033 ,  G10L21/00 ,  G10L2021/0135 ,  H04M3/51 ,  H04M2201/40

Abstract: A method and apparatus are provided for adjusting a content of an oral presentation provided by an agent of an organization and perceived by a human target of the organization based upon an objective of the organization. The method includes the steps of detecting a content of the oral presentation provided by the agent and modifying the oral presentation provided by the agent to produce the oral presentation perceived by the human target based upon the detected content and the organizational objective.

Abstract translation: 提供了一种方法和装置，用于基于组织的目标来调整由组织的代理提供并且由组织的人类目标所感知的口头呈现的内容。 该方法包括以下步骤：基于检测到的内容和组织目标，检测由代理提供的口头呈现的内容并修改由代理提供的口头表达以产生由人类目标感知的口头呈现。

公开(公告)号：US20060130637A1

公开(公告)日：2006-06-22

申请号：US10544189

申请日：2004-01-27

Applicant: Jean-Luc Crebouw

Inventor： Jean-Luc Crebouw

IPC: G10H7/00

CPC classification number: G10L21/0208 ,  G10L19/0204 ,  G10L2021/0135

Abstract: A method for differentiated digital voice and music processing, noise filtering and the creation of special effects. The method can be used to make the most of digital audio technologies, by performing a pre-encoding audio signal analysis, assuming that any sound signal during one frame interval is the sum of sines having a fixed amplitude and a frequency which is linearly modulated as a function of time, the sum being temporally modulated by the signal envelope and the noise being added to the signal prior to the sum.

Abstract translation: 一种用于区分数字语音和音乐处理，噪声过滤和创建特殊效果的方法。 该方法可以通过执行预编码音频信号分析来充分利用数字音频技术，假设在一个帧间隔期间的任何声音信号是具有固定振幅的正弦和被线性调制的频率之和 时间的函数，由信号包络暂时调制的和被加到和之前的信号的噪声。

公开(公告)号：US07003465B2

公开(公告)日：2006-02-21

申请号：US09975918

申请日：2001-10-12

Applicant: Keiko Morii ,  Yoshio Ohno

Inventor： Keiko Morii ,  Yoshio Ohno

IPC: G10L15/00

CPC classification number: G10L15/07 ,  G10L15/08 ,  G10L2021/0135

Abstract: An acoustic analysis unit acoustically analyzes a first utterance by a user. A pattern by-characteristic selection unit selects a trained pattern optimal for the user's utterance from a plurality of trained patterns that are previously classified and stored every characteristic. A speaker adaptation processor determines a spectral frequency distortion coefficient for correcting difference between spectral frequencies. The difference is caused by vocal tract length of a training speaker and an input speaker. Recognition of subsequent utterances using this determination improves recognition performance of the subsequent speech sounds.