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11. 发明授权

US5909663A Speech decoding method and apparatus for selecting random noise codevectors as excitation signals for an unvoiced speech frame 失效
标题翻译：用于选择随机噪声码矢量作为无声语音帧的激励信号的语音解码方法和装置
公开(公告)号：US5909663A
公开(公告)日：1999-06-01
申请号：US924142
申请日：1997-09-05
申请人： Kazuyuki Iijima , Masayuki Nishiguchi , Jun Matsumoto
发明人： Kazuyuki Iijima , Masayuki Nishiguchi , Jun Matsumoto
IPC分类号： G10L19/08 , G10L19/04 , H03M7/30 , G10L3/02
CPC分类号： G10L19/005
摘要： If the same parameter is repeatedly used in an unvoiced frame inherently devoid of pitch, there is produced a pitch of the frame length period, thus producing an extraneous feeling. This can be prevented from occurring by evading repeated use of excitation vectors having the same waveform shape. To this end, when decoding an encoded speech signal obtained on waveform encoding an encoding-unit-based time-axis speech signal obtained on splitting an input speech signal in terms of a pre-set encoding unit on the time axis, input data is checked by CRC by a CRC and bad frame masking circuit 281, which processes a frame corrupted with an error with bad frame masking of repeatedly using parameters of a directly previous frame. If the error-corrupted frame is unvoiced, an unvoiced speech synthesis unit 220 adds the noise to an excitation vector from a noise codebook or randomly selects the excitation vector of the noise codebook.
摘要翻译：如果相同的参数被重复使用在固有地没有间距的无声帧中，则产生帧长度周期的间距，从而产生无关紧要的感觉。可以通过避免重复使用具有相同波形形状的激励矢量来防止这种情况发生。为此，当对在时间轴上按预设编码单位分割输入语音信号而获得的基于编码单元的时间轴语音信号的波形编码获得的编码语音信号进行解码时，检查输入数据通过CRC通过CRC和坏帧屏蔽电路281，其处理使用直接前一帧的参数重复使用错误帧屏蔽的错误的帧。如果错误帧被清音，则无声语音合成单元220将噪声加到来自噪声码本的激励矢量，或者随机选择噪声码本的激励矢量。

12. 发明授权

US06611800B1 Vector quantization method and speech encoding method and apparatus 失效
标题翻译：矢量量化方法和语音编码方法及装置
公开(公告)号：US06611800B1
公开(公告)日：2003-08-26
申请号：US08927534
申请日：1997-09-11
申请人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto
发明人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto
IPC分类号： G10L1912
CPC分类号： H03M7/3082 , G10L19/0208 , G10L2019/0013
摘要： The processing volume for codebook search for vector quantization is diminished by sending data representing an envelope of spectral components of the harmonics from a spectrum evaluation unit 148 of a sinusoidal analytic encoder 114 to a vector quantizer 116 for vector quantization, so that the degree of similarity between an input vector and all code vectors stored in the codebook is found by approximation for pre-selecting a smaller number of code vectors. From these pre-selected code vectors, such a code vector minimizing an error with respect to the input vector is ultimately selected. In this manner, a smaller number of candidate code vectors are pre-selected by pre-selection involving simplified processing and subsequently subjected to ultimate selection with high precision.
摘要翻译：通过将表示谐波的频谱分量的包络的数据从正弦分析编码器114的频谱评估单元148发送到用于矢量量化的矢量量化器116来减少用于矢量量化的码本搜索的处理量，使得相似度通过近似来找到存储在码本中的输入向量和所有码矢量之间的预选择较少数量的码矢量。从这些预先选择的代码矢量中，最终选择最小化相对于输入向量的误差的码矢量。以这种方式，通过涉及简化处理的预选择来预先选择较少数量的候选代码矢量，并随后以高精度进行最终选择。

13. 发明授权

US6047253A Method and apparatus for encoding/decoding voiced speech based on pitch intensity of input speech signal 失效
标题翻译：基于输入语音信号的音调强度对有声语音进行编码/解码的方法和装置
公开(公告)号：US6047253A
公开(公告)日：2000-04-04
申请号：US925182
申请日：1997-09-08
申请人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto
发明人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto
IPC分类号： G10L11/04 , G06K7/10 , G10L11/02 , G10L11/06 , G10L19/00 , G10L19/04 , H03M7/30 , G10L9/14
CPC分类号： G10L25/93 , G10L19/093
摘要： A speech encoding method, a speech decoding method and corresponding apparatus capable of outputting non-buzzing spontaneous playback speech in a voiced portion includes a sinusoidal analysis encoding unit on the decoder side that detects the pitch of the voiced portion of the input speech signal. The pitch intensity information, which is a parameter containing the information representing not only the pitch intensity of the input speech signal but also the information representing proximity to the voiced speech or the unvoiced speech of the speech signal, is generated by a voiced/unvoiced (V/UV) discrimination unit and pitch intensity information generating circuit. The pitch intensity data is sent along with the encoded speech signal to the encoding side which then adds the noise component controlled on the basis of the pitch intensity information to the voiced portion of the encoded speech signal in a voiced speech synthesis portion and decodes and outputs the resulting signal.
摘要翻译：语音编码方法，语音解码方法以及能够在有声部分中输出非蜂鸣自发播放语音的对应装置包括：解码器侧的正弦分析编码单元，其检测输入语音信号的有声部分的音调。作为包含不仅表示输入语音信号的音高强度的信息的音调强度信息，也是表示语音信号的有声语音或无声语音的接近度的信息，由有声/无声（ V / UV）鉴别单元和音调强度信息产生电路。将音调强度数据与编码的语音信号一起发送到编码侧，然后编码侧将基于音高强度信息控制的噪声分量与有声语音合成部分中的编码语音信号的有声部分相加，并对其进行解码和输出产生的信号。

14. 发明授权

US06243672B1 Speech encoding/decoding method and apparatus using a pitch reliability measure 失效
标题翻译：语音编码/解码方法和使用音调可靠性度量的装置
公开(公告)号：US06243672B1
公开(公告)日：2001-06-05
申请号：US08927895
申请日：1997-09-11
申请人： Kazuyuki Iijima , Masayuki Nishiguchi , Jun Matsumoto
发明人： Kazuyuki Iijima , Masayuki Nishiguchi , Jun Matsumoto
IPC分类号： G10L1104
CPC分类号： G10L25/90
摘要： A pitch detection method and apparatus capable of realizing high-precision pitch detection even for speech signals in which half-pitch or double-pitch exhibits stronger autocorrelation than the pitch for detection. An input speech signal is judged as to voicedness or unvoicedness and a voiced portion and an unvoiced portion of the input speech signal are encoded by a sinusoidal analytic encoding unit 114 and by a code excitation encoding unit 120, respectively, for producing respective encoded outputs. The sinusoidal analytic encoding unit 114 performs pitch search on the encoded outputs for finding the pitch information from the input speech signal and sets the high-reliability pitch information based on the detected pitch information. The results of pitch detection are determined based on the high-reliability pitch information.
摘要翻译：即使对于半间距或双音调表现出比用于检测的音调更强的自相关性的语音信号，也能够实现高精度音调检测的音调检测方法和装置。输入语音信号被判断为浊音或清音，并且输入语音信号的有声部分和无声部分分别由正弦分析编码单元114和代码激励编码单元120编码，以产生相应的编码输出。正弦分析编码单元114对编码的输出执行音调搜索，以从输入语音信号中找出音调信息，并且基于检测到的音调信息设置高可靠性音调信息。基于高可靠性间距信息来确定音调检测的结果。

15. 发明授权

US6108621A Speech analysis method and speech encoding method and apparatus 失效
标题翻译：语音分析方法和语音编码方法及装置
公开(公告)号：US6108621A
公开(公告)日：2000-08-22
申请号：US946373
申请日：1997-10-07
申请人： Masayuki Nishiguchi , Jun Matsumoto , Kazuyuki Iijima , Akira Inoue
发明人： Masayuki Nishiguchi , Jun Matsumoto , Kazuyuki Iijima , Akira Inoue
IPC分类号： G10L11/04 , G10L19/02 , G10L19/04 , G10L19/08 , H04B14/04
CPC分类号： G10L19/08 , G10L19/10 , G10L25/90
摘要： A speech analysis method and a speech encoding method and apparatus in which, even if the harmonics of the speech spectrum are offset from integer multiples of the fundamental wave, the amplitudes of the harmonics can be evaluated correctly for producing a playback output of high clarity. To this end, the frequency spectrum of the input speech is split on the frequency axis into plural bands in each of which pitch search and evaluation of amplitudes of the harmonics are carried out simultaneously using an optimum pitch derived from the spectral shape. Using the structure of an harmonics as the spectral shape, and based on the rough pitch previously detected by an open-loop rough pitch search, a high-precision pitch search comprised of a first pitch search for the frequency spectrum in its entirety and a second pitch search of higher precision than the first pitch search is carried out. The second pitch search is performed independently for each of the high range side and the low range side of the frequency spectrum.
摘要翻译：一种语音分析方法和语音编码方法和装置，其中即使语音频谱的谐波偏离基波的整数倍，可以正确地评估谐波的幅度以产生高清晰度的播放输出。为此，输入语音的频谱在频率轴上被分割为多个频带，其中，使用从频谱形状导出的最佳音调同时进行音调的振幅搜索和评估。使用谐波的结构作为光谱形状，并且基于先前通过开环粗略音调搜索检测到的粗音调，包括对整个频谱进行第一音调搜索的高精度音调搜索和第二音调搜索执行比第一音调搜索更高精度的音调搜索。对于频谱的高范围侧和低范围侧的每一个独立地执行第二音调搜索。

16. 发明授权

US6018707A Vector quantization method, speech encoding method and apparatus 失效
标题翻译：矢量量化方法，语音编码方法和装置
公开(公告)号：US6018707A
公开(公告)日：2000-01-25
申请号：US924122
申请日：1997-09-05
申请人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto
发明人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto
IPC分类号： G10L15/02 , G10L19/00 , G10L19/02 , G10L19/04 , G10L19/08 , G10L19/14 , H03M7/30 , G10L9/14
CPC分类号： G10L19/02
摘要： The code vector search for vector-quantizing a variable-dimension input vector is to be improved in precision. Via a terminal are entered a variable number of data, that is a variable-dimension vector v, representing, for example, the amplitudes of spectral components of the harmonics of speech. The variable-dimension vector v is converted by a variable/fixed dimension conversion circuit into the vector x of a fixed dimension, such as 44-dimension vector, which is sent to a selection circuit. From plural fixed-dimension vectors, such a code vector as minimizes a weighted error is selected from a codebook. The code vector of fixed dimension obtained by the codebook is converted by a fixed/variable dimension converting circuit into the same variable dimension as that of the original variable-dimension vector v. The converted variable dimension code vector is sent to a variable-dimension selection circuit for selecting from the codebook such code vector as minimizes the weighted error from the input vector v.
摘要翻译：针对可变维度输入向量的向量量化的编码向量搜索的精度要提高。通过终端输入可变数量的数据，即可变维度向量+ E，uns v + EE，表示例如语音谐波的频谱分量的振幅。可变维度向量+ E，uns v + EE由可变/固定维度转换电路转换为固定维度的向量+ E，uns x + EE，例如44维向量，被发送到选择电路。从多个固定维度矢量中，从码本中选择最小化加权误差的码矢量。由码本获得的固定维数的代码矢量由固定/可变尺寸转换电路转换成与原始可变维度向量+ E，uns v + EE相同的可变维度。转换后的可变维码向量被发送到可变维选择电路，用于从码本中选择这样的代码向量，使得从输入向量+ E，uns v + EE最小化加权误差。

17. 发明授权

US06532443B1 Reduced length infinite impulse response weighting 失效
标题翻译：减少长度无限脉冲响应加权
公开(公告)号：US06532443B1
公开(公告)日：2003-03-11
申请号：US08951028
申请日：1997-10-15
申请人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto , Akira Inoue
发明人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto , Akira Inoue
IPC分类号： G10L1914
CPC分类号： G10L19/13 , G10L25/27
摘要： The processing volume in calculating a weight value for perceptually weighted vector quantization is decreased to speed up the processing or to minimize hardware. To this end, an inverted LPC finds LPC (linear prediction coding) residuals of an input speech signal which are processed with sinusoidal analysis encoding by a sinusoidal analysis encoding unit. The resulting parameters are processed by a vector quantizer with perceptually weighted vector quantization. For this perceptually weighted vector quantization, the weight value is calculated based on results of an orthogonal transform of parameters derived from the impulse response of the transfer function of the weight.
摘要翻译：在计算感知加权矢量量化的权重值时的处理量减少以加速处理或使硬件最小化。为此，反相LPC找到用正弦分析编码单元用正弦分析编码处理的输入语音信号的LPC（线性预测编码）残差。所得到的参数由感知加权矢量量化的矢量量化器处理。对于这种感知加权矢量量化，基于从权重的传递函数的脉冲响应导出的参数的正交变换的结果来计算权重值。

18. 发明授权

US06539355B1 Signal band expanding method and apparatus and signal synthesis method and apparatus 失效
标题翻译：信号带扩展方法及装置及信号合成方法及装置
公开(公告)号：US06539355B1
公开(公告)日：2003-03-25
申请号：US09417585
申请日：1999-10-14
申请人： Shiro Omori , Masayuki Nishiguchi
发明人： Shiro Omori , Masayuki Nishiguchi
IPC分类号： G10L1902
CPC分类号： G10L21/038
摘要： A bandwidth expanding method and apparatus in which frequency characteristics of high-frequency components of broad band signals can be adjusted to the liking of the user, overflow due to addition is prevented from occurring without power variations being perceived by a user, the number of broad band formants is reduced, and emphasis is attached to the rough structure of the spectrum, so that the produced broad band speech signals can be improved in quality. To this end, in a speech bandwidth expansion device, frequency characteristics of the frequency components not less than 3400 Hz are adjusted by preset alterable parameter values and summed to the original narrow band speech components. If overflow has occurred in a sample, the high-range gain of the sample is lowered to a level below the overflow level before proceeding to addition. Also, broad band autocorrelation &ggr;w is generated and inverse-transformed in an inverse parameter conversion unit to produce broad band linear prediction coefficient &agr;W to synthesize the broad-band speech in a linear predictive coding synthesis unit.
摘要翻译：宽带信号的高频分量的频率特性可以根据用户的喜好进行调整的带宽扩展方法和装置，防止由于添加而导致的溢出，而不会由用户感知到功率变化，广泛的数量频带共振峰减少，重点在于光谱的粗糙结构，从而可以提高产生的宽带语音信号的质量。为此，在语音带宽扩展装置中，频率分量不小于3400Hz的频率特性通过预设的可变参数值进行调整，并与原始窄带语音分量相加。如果在样品中发生溢出，则在继续添加之前，将样品的高范围增益降低到低于溢出水平的水平。此外，在逆参数转换单元中产生宽带自相关法拉姆并逆变换，以产生宽带线性预测系数αW，以在线性预测编码合成单元中合成宽带语音。

19. 发明授权

US06732075B1 Sound synthesizing apparatus and method, telephone apparatus, and program service medium 失效
标题翻译：声音合成装置和方法，电话设备和程序服务介质
公开(公告)号：US06732075B1
公开(公告)日：2004-05-04
申请号：US09556036
申请日：2000-04-20
申请人： Shiro Omori , Masayuki Nishiguchi
发明人： Shiro Omori , Masayuki Nishiguchi
IPC分类号： G10L1700
CPC分类号： G10L19/012 , G10L21/038
摘要： In a sound synthesizer, a noise adder generates a noise signal having a frequency band of 3,400 to 4,600 Hz, adjusts the gain of the noise signal, and adds the gain-adjusted noise signal to an excitation source after being filled with zeros by a zero-filling circuit, thereby providing a wide-band excitation source which is rather flat. The signal gain is adjusted by determining a narrow-band excitation source or a power of the wide-band excitation source after being filled with zeros and fitting the gain to the narrow-band excitation source or the power.
摘要翻译：在声音合成器中，噪声加法器产生具有3,400至4,600Hz的频带的噪声信号，调整噪声信号的增益，并将经增益调整的噪声信号加到零点之后的激励源上，零填充电路，从而提供相当平坦的宽带激发源。信号增益通过在填充零之后确定窄带激发源或宽带激发源的功率并将增益拟合到窄带激发源或功率来调整。

20. 发明授权

US06289311B1 Sound synthesizing method and apparatus, and sound band expanding method and apparatus 失效
标题翻译：声音合成方法和装置，以及声带扩展方法和装置
公开(公告)号：US06289311B1
公开(公告)日：2001-09-11
申请号：US09175616
申请日：1998-10-20
申请人： Shiro Omori , Masayuki Nishiguchi
发明人： Shiro Omori , Masayuki Nishiguchi
IPC分类号： G10L1302
CPC分类号： G10L21/038 , G10L25/93 , G10L2019/0005
摘要： A method and apparatus for sound synthesizing and sound band expanding of a narrow band input signal uses wide-band voiced and unvoiced sound code books and also uses narrow-band voiced and unvoiced sound code books. Coded input sound parameters are decoded and quantized using the narrow-band voiced and unvoiced sound code books and are then de-quantized using the wide-band voiced and unvoiced sound code books. The sound is synthesized based on the de-quantized data and a so-called innovation-related parameter formed by a zero-filling circuit filing zeros between samples of the framed input signal, so that the result is an upsampled aliased wide-band signal used with the de-quantized data to synthesize the sound.
摘要翻译：用于窄带输入信号的声合成和声带扩展的方法和装置使用宽带有声和无声声码簿，并且还使用窄带浊音和无声声码书。编码输入声音参数使用窄带有声和无声声码书进行解码和量化，然后使用宽带有声和无声声码簿进行去量化。该声音是基于去量化的数据和所谓的创新相关参数合成的，该参数由零填充电路形成，在成帧的输入信号的样本之间归零，从而得到结果是上采样的混叠宽带信号使用去量化的数据来合成声音。

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