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1. 发明申请

US20120065980A1 CODING AND DECODING A TRANSIENT FRAME 有权
标题翻译：编码和解码一个瞬态帧
公开(公告)号：US20120065980A1
公开(公告)日：2012-03-15
申请号：US13228210
申请日：2011-09-08
申请人： Venkatesh Krishnan , Ananthapadmanabhan Arasanipalai Kandhadai
发明人： Venkatesh Krishnan , Ananthapadmanabhan Arasanipalai Kandhadai
IPC分类号： G10L13/00
CPC分类号： G10L19/20 , G10L19/025 , G10L19/097 , G10L19/22 , G10L25/93
摘要： An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.
摘要翻译：描述用于对瞬态帧进行编码的电子设备。电子设备包括处理器和存储在与处理器电子通信的存储器中的可执行指令。电子设备获得当前瞬态帧。电子设备还基于当前瞬态帧获得残留信号。另外，电子设备基于剩余信号确定一组峰值位置。电子设备还基于至少一组峰值位置来确定是否使用第一编码模式或第二编码模式来编码当前瞬态帧。如果确定了第一编码模式，则电子设备还基于第一编码模式合成激励。如果确定了第二编码模式，则电子设备还基于第二编码模式合成激励。

2. 发明申请

US20110087488A1 SPEECH SYNTHESIS APPARATUS AND METHOD 有权
标题翻译：语音合成设备和方法
公开(公告)号：US20110087488A1
公开(公告)日：2011-04-14
申请号：US12970162
申请日：2010-12-16
申请人： Ryo Morinaka , Takehiko Kagoshima
发明人： Ryo Morinaka , Takehiko Kagoshima
IPC分类号： G10L11/04 , G10L13/06
CPC分类号： G10L13/06 , G10L13/033 , G10L19/097 , G10L25/15 , G10L2021/0135
摘要： According to an embodiment, a speech synthesis apparatus includes a selecting unit configured to select speaker's parameters one by one for respective speakers and obtain a plurality of speakers' parameters, the speaker's parameters being prepared for respective pitch waveforms corresponding to speaker's speech sounds, the speaker's parameters including formant frequencies, formant phases, formant powers, and window functions concerning respective formants that are contained in the respective pitch waveforms. The apparatus includes a mapping unit configured to make formants correspond to each other between the plurality of speakers' parameters using a cost function based on the formant frequencies and the formant powers. The apparatus includes a generating unit configured to generate an interpolated speaker's parameter by interpolating, at desired interpolation ratios, the formant frequencies, formant phases, formant powers, and window functions of formants which are made to correspond to each other.
摘要翻译：根据实施例，语音合成装置包括：选择单元，被配置为逐个选择说话者的参数，并且获得多个扬声器的参数;所述说话者的参数是针对对应于说话者的语音的各个音调波形而准备的，参数包括共振峰频率，共振峰相位，共振峰功率，以及相关螺旋波形中包含的各共振峰的窗函数。该装置包括：映射单元，其被配置为使用基于共振峰频率和共振峰功率的成本函数在多个扬声器的参数之间使得共振峰彼此对应。该装置包括：生成单元，被配置为通过以期望的内插比率内插使彼此对应的共振峰的共振峰频率，共振峰相位，共振峰功率和窗函数来生成内插说话者的参数。

3. 发明申请

US20080312917A1 METHOD AND APPARATUS FOR PREDICTIVELY QUANTIZING VOICED SPEECH 有权
标题翻译：用于预测定语音的方法和装置
公开(公告)号：US20080312917A1
公开(公告)日：2008-12-18
申请号：US12190524
申请日：2008-08-12
申请人： Arasanipalai K. Ananthapadmanabhan , Sharath Manjunath , Pengjun Huang , Eddie-Lun Tik Choy , Andrew P. Dejaco
发明人： Arasanipalai K. Ananthapadmanabhan , Sharath Manjunath , Pengjun Huang , Eddie-Lun Tik Choy , Andrew P. Dejaco
IPC分类号： G10L19/00
CPC分类号： G10L19/04 , G10L19/0204 , G10L19/032 , G10L19/08 , G10L19/097 , G10L19/26 , G10L25/12
摘要： A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.
摘要翻译：用于预测量化浊音的方法和装置包括参数发生器和量化器。参数发生器被配置为从诸如有声语音的预测语音的帧中提取参数，并且将提取的信息变换为频域表示。量化器被配置为从当前帧的参数中减去先前帧的参数的加权和。量化器被配置为量化差值。可以添加原型提取器以首先提取要由参数发生器处理的音调周期原型。

4. 发明申请

US20070271091A1 Apparatus, method and program for vioce signal interpolation 有权
标题翻译：用于vioce信号插值的装置，方法和程序
公开(公告)号：US20070271091A1
公开(公告)日：2007-11-22
申请号：US11797701
申请日：2007-05-07
申请人： Yasushi Sato
发明人： Yasushi Sato
IPC分类号： G10L11/04
CPC分类号： G10L21/0364 , G10L19/09 , G10L19/097 , G10L25/18
摘要： A voice signal interpolation apparatus is provided which can restore original human voices from human voices in a compressed state while maintaining a high sound quality. When a voice signal representative of a voice to be interpolated is acquired by a voice data input unit 1, a pitch deriving unit 2 filters this voice signal to identify a pitch length from the filtering result. A pitch length fixing unit 3 makes the voice signal have a constant time length of a section corresponding to a unit pitch, and generates pitch waveform data. A sub-band dividing unit 4 converts the pitch waveform data into sub-band data representative of a spectrum. A plurality of sub-band data pieces are averaged by an averaging unit 5 and thereafter a sub-band synthesizing unit 6 converts the sub-band data pieces into a signal representative of a waveform of the voice by a sub-band synthesizing unit 6. The time length of this signal in each section is restored by a pitch restoring unit 7 and a sound output unit 8 reproduces the sound represented by the signal.
摘要翻译：提供了一种语音信号插值装置，其能够在保持高音质的同时在压缩状态下恢复来自人声音的原始人声。当语音数据输入单元1获取表示要内插的语音的语音信号时，音调导出单元2对该语音信号进行滤波，以从滤波结果中识别音调长度。音高固定单元3使语音信号具有与单位音调对应的部分的恒定时间长度，并产生音调波形数据。子带分割单元4将音调波形数据转换为表示频谱的子带数据。多个子带数据由平均单元5进行平均，此后，子带合成单元6将子带数据段转换成表示子频带合成单元6的声音波形的信号。每个部分中的该信号的时间长度由音调恢复单元7恢复，并且声音输出单元8再现由该信号表示的声音。

5. 发明授权

US07085712B2 Method and apparatus for subsampling phase spectrum information 有权
公开(公告)号：US07085712B2
公开(公告)日：2006-08-01
申请号：US10702967
申请日：2003-11-05
申请人： Sharath Manjunath
发明人： Sharath Manjunath
IPC分类号： G10L19/02
CPC分类号： G10L19/097 , G10L19/02 , G10L25/27
摘要： Method and apparatus for subsampling phase spectrum information by analyzing and reconstructing a prototype of a frame. The prototype is analyzed by correlating phase parameters generated from the prototype with phase parameters generated from a reference prototype in multiple frequency bands. The prototype is reconstructed using linear phase shift values by producing a set of phase parameters of the reference prototype, generating a set of linear phase shift values associated with the prototype, and composing a phase vector from the set of phase parameters and the set of linear phase shift values across multiple frequency bands. The prototype is reconstructed using circular rotation values by producing a set of circular rotation values associated with the prototype, generating a set of bandpass waveforms associated with the phase parameters of the reference prototype in multiple frequency bands, and modifying the set of bandpass waveforms based upon the circular rotation values.

6. 发明授权

US06801887B1 Speech coding exploiting the power ratio of different speech signal components 失效
标题翻译：语音编码利用不同语音信号分量的功率比
公开(公告)号：US06801887B1
公开(公告)日：2004-10-05
申请号：US09666971
申请日：2000-09-20
申请人： Ari Heikkinen , Mikko Tammi , Jani Nurminen
发明人： Ari Heikkinen , Mikko Tammi , Jani Nurminen
IPC分类号： G10L1914
CPC分类号： G10L19/097 , G10L19/24
摘要： A method and system for waveform interpolation speech coding. The method comprises the steps of decomposing the speech signal into a slowly evolving waveform component and a rapidly evolving waveform component in the encoder and determining the power ratio of these surface components so that the power ratio can be used to determine the bit allocation when the surface components are quantized. The power ratio can also be used to modify the phases of the slowly evolving waveform component when the surface components are reconstructed in the decoder in order to improve the speech quality.
摘要翻译：一种用于波形插值语音编码的方法和系统。该方法包括以下步骤：将语音信号分解成编码器中缓慢演变的波形分量和快速演变的波形分量，并确定这些表面分量的功率比，使得当表面的比特分配时可以使用功率比来确定比特分配组分被量化。当在解码器中重构表面分量以便改善语音质量时，功率比也可用于修改缓慢演变的波形分量的相位。

7. 发明申请

US20040153314A1 Speech signal interpolation device, speech signal interpolation method, and program 有权
标题翻译：语音信号插值装置，语音信号插补方法和程序
公开(公告)号：US20040153314A1
公开(公告)日：2004-08-05
申请号：US10477320
申请日：2003-11-10
发明人： Yasushi Sato
IPC分类号： G10L011/04
CPC分类号： G10L21/0364 , G10L19/09 , G10L19/097 , G10L25/18
摘要： A voice signal interpolation apparatus is provided which can restore original human voices from human voices in a compressed state while maintaining a high sound quality. When a voice signal representative of a voice to be interpolated is acquired by a voice data input unit 1, a pitch deriving unit 2 filters this voice signal to identify a pitch length from the filtering result. A pitch length fixing unit 3 makes the voice signal have a constant time length of a section corresponding to a unit pitch, and generates pitch waveform data. A sub-band dividing unit 4 converts the pitch waveform data into sub-band data representative of a spectrum. A plurality of sub-band data pieces are averaged by an averaging unit 5 and thereafter a sub-band synthesizing unit 6 converts the sub-band data pieces into a signal representative of a waveform of the voice by a sub-band synthesizing unit 6. The time length of this signal in each section is restored by a pitch restoring unit 7 and a sound output unit 8 reproduces the sound represented by the signal.
摘要翻译：提供了一种语音信号插值装置，其能够在保持高音质的同时在压缩状态下恢复来自人声音的原始人声。当语音数据输入单元1获取表示要内插的语音的语音信号时，音调导出单元2对该语音信号进行滤波，以从滤波结果中识别音调长度。音高固定单元3使语音信号具有与单位音调对应的部分的恒定时间长度，并产生音调波形数据。子带分割单元4将音调波形数据转换为表示频谱的子带数据。多个子带数据由平均单元5进行平均，此后，子带合成单元6将子带数据段转换成表示子频带合成单元6的声音波形的信号。每个部分中的该信号的时间长度由音调恢复单元7恢复，并且声音输出单元8再现由该信号表示的声音。

8. 发明申请

US20040098431A1 Device and method for interpolating frequency components of signal 有权
标题翻译：用于内插信号频率分量的装置和方法
公开(公告)号：US20040098431A1
公开(公告)日：2004-05-20
申请号：US10362421
申请日：2003-02-25
发明人： Yasushi Sato
IPC分类号： G06F007/38
CPC分类号： G10L21/038 , G10L19/0204 , G10L19/097 , H04B1/667
摘要： A frequency interpolation apparatus is provided which reproduces a signal similar to an original signal by approximately recovering suppressed frequency components, from an input signal having the suppressed frequency components in a specific frequency band of the original signal. The input signal is divided into a plurality of signal component sets each having frequency components in a frequency band among a plurality of frequency bands, and a signal component set in the band with the suppressed signal components is synthesized from the plurality of divided signal component sets and added to the input signal. Each of the plurality of divided signal component sets is frequency-converted to a signal component set in the same frequency band, and the signal component set in the band with the suppressed signal components is synthesized through linear combination of the frequency-converted signal component sets. Spectrum envelope information of the frequency components not suppressed but residual in the original signal is extracted and the level of the signal component set to be synthesized is determined from the spectrum envelope information.
摘要翻译：提供了一种频率内插装置，其从原始信号的特定频带中具有抑制的频率分量的输入信号中大致恢复抑制的频率分量，再现与原始信号类似的信号。输入信号被分成多个信号分量集合，每个信号分量集合具有多个频带中的频带中的频率分量，并且从多个分割信号分量集合合成具有抑制信号分量的频带中设置的信号分量并添加到输入信号。多个分割信号分量集合中的每一个被频率转换成在相同频带中设置的信号分量，并且通过频率转换信号分量集合的线性组合来合成具有抑制信号分量的频带中设置的信号分量。从频谱包络信息中提取未抑制但原始信号中的残差的频率分量的频谱包络信息，并确定要合成的信号分量的电平。

9. 发明授权

US06678649B2 Method and apparatus for subsampling phase spectrum information 有权
标题翻译：二次采样相位谱信息的方法和装置
公开(公告)号：US06678649B2
公开(公告)日：2004-01-13
申请号：US10066073
申请日：2002-02-01
申请人： Sharath Manjunath
发明人： Sharath Manjunath
IPC分类号： G10L1912
CPC分类号： G10L19/097 , G10L19/02 , G10L25/27
摘要： Method and apparatus for subsampling phase spectrum information by analyzing and reconstructing a prototype of a frame. The prototype is analyzed by correlating phase parameters generated from the prototype with phase parameters generated from a reference prototype in multiple frequency bands. The prototype is reconstructed using linear phase shift values by producing a set of phase parameters of the reference prototype, generating a set of linear phase shift values associated with the prototype, and composing a phase vector from the set of phase parameters and the set of linear phase shift values across multiple frequency bands. The prototype is reconstructed using circular rotation values by producing a set of circular rotation values associated with the prototype, generating a set of bandpass waveforms associated with the phase parameters of the reference prototype in multiple frequency bands, and modifying the set of bandpass waveforms based upon the circular rotation values.
摘要翻译：通过分析和重建帧的原型对相位频谱信息进行子采样的方法和装置。通过将从原型产生的相位参数与在多个频带中的参考原型生成的相位参数相关联来分析原型。使用线性相移值通过产生参考原型的一组相位参数来重构原型，产生与原型相关联的一组线性相移值，以及从相位参数集合和线性组合构成相位矢量跨越多个频带的相移值。通过产生与原型相关联的一组圆形旋转值，使用循环旋转值重构原型，产生与多个频带中的参考原型的相位参数相关联的一组带通波形，以及基于循环旋转值。

10. 发明申请

US20020116184A1 REW parametric vector quantization and dual-predictive SEW vector quantization for waveform interpolative coding 有权
标题翻译： REW参数矢量量化和用于波形内插编码的双重预测SEW矢量量化
公开(公告)号：US20020116184A1
公开(公告)日：2002-08-22
申请号：US09811187
申请日：2001-03-16
发明人： Oded Gottsman , Allen Gersho
IPC分类号： G10L019/10
CPC分类号： G10L19/097
摘要： An enhanced analysis-by-synthesis waveform interpolative speech coder able to operate at 2.8 kbps. Novel features include dual-predictive analysis-by-synthesis quantization of the slowly-evolving waveform, efficient parametrization of the rapidly-evolving waveform magnitude, and analysis-by-synthesis vector quantization of the rapidly evolving waveform parameter. Subjective quality tests indicate that it exceeds G.723.1 at 5.3 kbps, and of G.723.1 at 6.3 kbps.
摘要翻译：增强的综合波形内插语音编码器能够以2.8 kbps的速率工作。新颖的功能包括缓慢演变的波形的双重预测分析合成量化，快速演化的波形幅度的有效参数化以及快速演变的波形参数的按合成矢量量化。主观质量测试表明，它在5.3 kbps处超过G.723.1，在6.3kbps处超过G.723.1。

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