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1. 发明授权

US5381482A Sound field controller 失效
标题翻译：声场控制器
公开(公告)号：US5381482A
公开(公告)日：1995-01-10
申请号：US12265
申请日：1993-02-01
申请人： Masaharu Matsumoto , Mitsuhiko Serikawa , Akihisa Kawamura , Hiroko Numazu , Takeshi Norimatsu , Ryo Tagami , Mikio Oda
发明人： Masaharu Matsumoto , Mitsuhiko Serikawa , Akihisa Kawamura , Hiroko Numazu , Takeshi Norimatsu , Ryo Tagami , Mikio Oda
IPC分类号： H04S1/00 , H04S3/00 , H04S5/02 , H04S7/00
CPC分类号： H04S1/002 , H04S7/305 , H04S1/007 , H04S2400/01 , H04S5/00
摘要： A sound field controller for generating apparent sound sources by adjusting the amplitude and delay time of a sound signal so that the sound will be perceived by plural listeners as sound coming from a location separated from the specific location of the front speakers, and for additionally controlling the effect of the apparent sound sources by evaluating the attributes of the source sound signal. The controller includes FIR filters for generating a left sound pattern signal, FIR filters for generating a right sound pattern signal, a first delay circuit for delaying the left and right sound pattern signals by a first predetermined time and applying the delayed left and right sound pattern signals to the left and right speakers, respectively, to introduce an apparent sound source located left rear of a center listener; and a second delay circuit for delaying the left and right sound pattern signals by a second predetermined time and applying the delayed left and right sound pattern signals to the right and left speakers, respectively, to introduce an apparent sound source located right rear of a center listener.
摘要翻译：一种声场控制器，用于通过调节声音信号的幅度和延迟时间来产生视在声源，使得声音将被多个听众感知为来自与前置扬声器的特定位置分离的位置的声音，并且用于另外控制通过评估源声音信号的属性来观察视声源的影响。控制器包括用于产生左声音模式信号的FIR滤波器，用于产生右声音模式信号的FIR滤波器，用于将左右声音图形信号延迟第一预定时间的第一延迟电路，并将延迟的左右声音模式分别向左右扬声器发出信号，引入位于中心听众左后方的明显声源; 以及第二延迟电路，用于将左右声音图形信号延迟第二预定时间，并将延迟的左右声音图案信号分别施加到左右扬声器，以引入位于中心右后方的表观声源听众

2. 发明授权

US5617478A Sound reproduction system and a sound reproduction method 失效
标题翻译：声音再现系统和声音再现方法
公开(公告)号：US5617478A
公开(公告)日：1997-04-01
申请号：US225602
申请日：1994-04-11
申请人： Ryou Tagami , Takeshi Norimatsu , Masaharu Matsumoto , Mikio Oda , Mitsuhiko Serikawa , Akihisa Kawamura , Hiroko Numazu
发明人： Ryou Tagami , Takeshi Norimatsu , Masaharu Matsumoto , Mikio Oda , Mitsuhiko Serikawa , Akihisa Kawamura , Hiroko Numazu
IPC分类号： H03G5/00 , H04R5/04 , H04R29/00 , H03G3/00
CPC分类号： H04R5/04 , H03G5/005
摘要： A sound reproduction system for automatically conducting a tone control for an audio signal according to the present invention includes: an input terminal for inputting an audio signal; a judgment circuit for determining a ratio between a stereophonic signal component and a monophonic signal component of the input audio signal and for outputting a judgment signal; a signal processing section for receiving the input audio signal and processing the input audio signal so as to generate an output signal having uniform transmission characteristics irrespective of the listening position; an adder for receiving the input audio signal and the output signal and for adding the input audio signal with the output signal at a certain addition ratio based on the judgment signal so as to generate an added signal; and a loudspeaker for receiving the added signal and for reproducing the added signal in a plurality of positions.
摘要翻译：根据本发明的用于自动进行音频信号的音调控制的声音再现系统包括：用于输入音频信号的输入端; 用于确定所述输入音频信号的立体声信号分量和单声道信号分量之间的比率并用于输出判断信号的判断电路; 信号处理部分，用于接收输入音频信号并处理输入音频信号，以产生具有均匀传输特性的输出信号，而与收听位置无关; 加法器，用于接收所述输入音频信号和所述输出信号，并且用于基于所述判断信号以一定的相加比例将所述输入音频信号与所述输出信号相加以产生相加信号; 以及用于接收所附加的信号并用于在多个位置再现附加的信号的扬声器。

3. 发明授权

US06477490B2 Audio signal compression method, audio signal compression apparatus, speech signal compression method, speech signal compression apparatus, speech recognition method, and speech recognition apparatus 有权
标题翻译：音频信号压缩方法，音频信号压缩装置，语音信号压缩方法，语音信号压缩装置，语音识别方法和语音识别装置
公开(公告)号：US06477490B2
公开(公告)日：2002-11-05
申请号：US09892745
申请日：2001-06-28
申请人： Yoshihisa Nakatoh , Takeshi Norimatsu , Mineo Tsushima , Tomokazu Ishikawa , Mitsuhiko Serikawa , Taro Katayama , Junichi Nakahashi , Yoriko Yagi
发明人： Yoshihisa Nakatoh , Takeshi Norimatsu , Mineo Tsushima , Tomokazu Ishikawa , Mitsuhiko Serikawa , Taro Katayama , Junichi Nakahashi , Yoriko Yagi
IPC分类号： G10L1906
CPC分类号： H04B1/665 , G10L2019/0005
摘要： An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要翻译：一种用于对输入音频信号进行压缩编码的音频信号压缩装置包括用于将输入音频信号变换为频域信号的时间 - 频率变换单元; 频谱包络计算单元，用于根据输入的音频信号，使用基于人的听觉特征的频率的加权函数来计算用于不同频率的不同分辨率的频谱包络; 归一化单元，用于使用频谱包络对频域信号进行归一化以获得残余信号; 功率归一化单元，用于通过所述功率归一化所述残余信号; 听觉加权计算单元，用于基于输入音频信号的频谱和人类听觉特征来计算频率上的加权系数; 以及具有串联连接的多级矢量量化器的多级量化装置，其中输入归一化残差信号，以及使用加权系数量化残差信号的矢量量化器中的至少一个。因此，与高频带相比，可以以更高的频率分辨率来分析具有重要意义的低频带，从而实现利用人类听觉特性的有效信号压缩。

4. 发明授权

US06311153B1 Speech recognition method and apparatus using frequency warping of linear prediction coefficients 有权
标题翻译：使用线性预测系数的频率变形的语音识别方法和装置
公开(公告)号：US06311153B1
公开(公告)日：2001-10-30
申请号：US09165297
申请日：1998-10-02
申请人： Yoshihisa Nakatoh , Takeshi Norimatsu , Mineo Tsushima , Tomokazu Ishikawa , Mitsuhiko Serikawa , Taro Katayama , Junichi Nakahashi , Yoriko Yagi
发明人： Yoshihisa Nakatoh , Takeshi Norimatsu , Mineo Tsushima , Tomokazu Ishikawa , Mitsuhiko Serikawa , Taro Katayama , Junichi Nakahashi , Yoriko Yagi
IPC分类号： G01L2100
CPC分类号： H04B1/665 , G10L2019/0005
摘要： An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要翻译：一种用于对输入音频信号进行压缩编码的音频信号压缩装置包括用于将输入音频信号变换为频域信号的时间 - 频率变换单元; 频谱包络计算单元，用于根据输入的音频信号，使用基于人的听觉特征的频率的加权函数来计算用于不同频率的不同分辨率的频谱包络; 归一化单元，用于使用频谱包络对频域信号进行归一化以获得残余信号; 功率归一化单元，用于通过所述功率归一化所述残余信号; 听觉加权计算单元，用于基于输入音频信号的频谱和人类听觉特征来计算频率上的加权系数; 以及具有串联连接的多级矢量量化器的多级量化装置，其中输入归一化残差信号，以及使用加权系数量化残差信号的矢量量化器中的至少一个。因此，与高频带相比，可以以更高的频率分辨率来分析具有重要意义的低频带，从而实现利用人类听觉特性的有效信号压缩。

5. 发明授权

US08284961B2 Signal processing device 有权
标题翻译：信号处理装置
公开(公告)号：US08284961B2
公开(公告)日：2012-10-09
申请号：US11995571
申请日：2006-07-10
申请人： Shuji Miyasaka , Yosiaki Takagi , Takeshi Norimatsu , Akihisa Kawamura , Kojiro Ono
发明人： Shuji Miyasaka , Yosiaki Takagi , Takeshi Norimatsu , Akihisa Kawamura , Kojiro Ono
IPC分类号： H04B1/00
CPC分类号： H04S3/002 , G10L19/008 , H04S2420/03
摘要： A signal processing device includes a generation unit that generates a second signal from a first signal that is obtained by down mixing two signals; a mixing coefficient determination unit that determines, based on a value L and a value θ, a mixing degree for mixing the first signal and the second signal; and a mixing unit that mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit. The generation unit includes a first filter that generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter that generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter is a filter unit which, for a complex-number signal, de-correlates an input signal and adds a reverberation component by using a delay unit and an all pass filter, and the processing unit is a filter unit different from the first filter.
摘要翻译：信号处理装置包括生成单元，该生成单元从通过混合两个信号获得的第一信号产生第二信号; 混合系数确定单元，其基于值L和值来确定用于混合第一信号和第二信号的混合度; 以及混合单元，其基于由混合系数确定单元确定的混合程度来混合第一信号和第二信号。该生成单元包括从第一信号中的低频带信号在第二信号中产生低频带信号的第一滤波器; 以及第二滤波器，其从第一信号中的高频带信号在第二信号中产生高频带信号。第一滤波器是滤波器单元，对于复数信号，使输入信号去相关，并通过使用延迟单元和全通滤波器来添加混响分量，并且处理单元是与第一滤波器不同的滤波器单元过滤。

6. 发明申请

US20090122182A1 SIGNAL PROCESSING DEVICE 有权
标题翻译：信号处理装置
公开(公告)号：US20090122182A1
公开(公告)日：2009-05-14
申请号：US11995571
申请日：2006-07-10
申请人： Shuji Miyasaka , Yosiaki Takagi , Takeshi Norimatsu , Akihisa Kawamura , Kojiro Ono
发明人： Shuji Miyasaka , Yosiaki Takagi , Takeshi Norimatsu , Akihisa Kawamura , Kojiro Ono
IPC分类号： H04N7/12
CPC分类号： H04S3/002 , G10L19/008 , H04S2420/03
摘要： A signal processing device (1) includes: a generation unit (32) which generates a second signal from a first signal that is obtained by downmixing two signals; a mixing coefficient determination unit (40) which determines, based on a value L and a value θ, a mixing degree for mixing the first signal and the second signal, the value L indicating a level ratio between the two signals, and the value θ indicating a phase difference between the two signals; and a mixing unit (50) which mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit (40). The generation unit (32) includes: a first filter (302) which generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter (a processing unit 307) which generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter (302) is a filter unit which, for a complex-number signal, decorrelates an input signal and adds a reverberation component by using a delay unit (301) and an all pass filter, and the processing unit (307) is a filter unit different from the first filter (302).
摘要翻译：信号处理装置（1）包括：生成单元（32），其从通过对两个信号进行下混合获得的第一信号产生第二信号; 混合系数确定单元（40），其基于值L和值θ来确定用于混合第一信号和第二信号的混合度，表示两个信号之间的电平比的值L和值θ 表示两个信号之间的相位差; 以及基于由混合系数确定单元（40）确定的混合程度混合第一信号和第二信号的混合单元（50）。生成单元（32）包括：从第一信号中的低频带信号生成第二信号中的低频带信号的第一滤波器（302）以及从第一信号中的高频带信号在第二信号中产生高频带信号的第二滤波器（处理单元307）。第一滤波器（302）是滤波器单元，对于复数信号，通过使用延迟单元（301）和全通滤波器去除相关输入信号并添加混响分量，并且处理单元（307）是与第一过滤器（302）不同的过滤器单元。

7. 发明授权

US08081764B2 Audio decoder 有权
标题翻译：音频解码器
公开(公告)号：US08081764B2
公开(公告)日：2011-12-20
申请号：US11993066
申请日：2006-07-11
申请人： Yosiaki Takagi , Kok seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono
发明人： Yosiaki Takagi , Kok seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono
IPC分类号： H04R5/00
CPC分类号： G10L19/008 , G10L19/0204
摘要： Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder (102) and an analysis filter bank (110) which generate, from a coded down-mixed signal, the first frequency band signal (x) corresponding to a down-mixed signal (M); a channel expansion unit (130) which converts the first frequency band signal (x) generated by the analysis filter bank (110) into output signals (y) corresponding to respective audio signals of N channels, using BC information; an synthesis filter bank (140) which performs band synthesis for the output signals (y) generate by the channel expansion unit (130) and thereby converts the output signals (y) into the respective audio signals of the N channels on a time axis; and an aliasing noise detection unit (120) which detects occurrence of aliasing noise in the first frequency band signal (x). The channel expansion unit (130) further prevents the aliasing noise from being included in the output signals (y), based on information detected by the aliasing noise detection unit (120).
摘要翻译：提供一种音频解码器，其可以在抑制混叠噪声的发生的同时减少算术运算量。音频解码器包括：解码器（102）和分析滤波器组（110），其从编码的下变频信号产生与下混合信号（M）对应的第一频带信号（x）; 信道扩展单元，其使用BC信息将由分析滤波器组（110）生成的第一频带信号（x）转换成对应于N个信道的各个音频信号的输出信号（y）; 合成滤波器组（140），其对由所述信道扩展单元（130）生成的输出信号（y）进行频带合成，从而将输出信号（y）转换成时间轴上的N个信道的各个音频信号; 以及用于检测第一频带信号（x）中的混叠噪声的出现的混叠噪声检测单元（120）。信道扩展单元（130）还基于混叠噪声检测单元（120）检测到的信息进一步防止混叠噪声包含在输出信号（y）中。

8. 发明授权

US08019614B2 Energy shaping apparatus and energy shaping method 有权
标题翻译：能量整形设备和能量整形方法
公开(公告)号：US08019614B2
公开(公告)日：2011-09-13
申请号：US12065378
申请日：2006-08-31
申请人： Yoshiaki Takagi , Kok Seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono , Tomokazu Ishikawa
发明人： Yoshiaki Takagi , Kok Seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono , Tomokazu Ishikawa
IPC分类号： G10L19/00
CPC分类号： G10L19/26 , G10L19/008 , G10L19/0204 , H04S2420/03
摘要： A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.
摘要翻译：时间处理装置包括：分离器，将包括在子带域中的音频信号分成指示混响分量的漫射信号和指示非混响分量的直接信号; 下混合单元，通过使直接信号下混合来产生降混信号; BPF分别产生带通下混信号和带通漫射信号; 归一化处理单元，分别产生归一化的下混信号和归一化扩散信号; 比例计算处理单元在预定时隙的基础上计算指示归一化的下混信号相对于归一化扩散信号的能量的能量的大小的比例因子; 计算单元，生成缩放漫射信号; HPF产生高通漫反射信号; 添加单元生成附加信号; 以及合成滤波器组，对加法信号执行合成滤波处理，并将加法信号转换成时域。

9. 发明授权

US08184817B2 Multi-channel acoustic signal processing device 有权
标题翻译：多通道声信号处理装置
公开(公告)号：US08184817B2
公开(公告)日：2012-05-22
申请号：US12064975
申请日：2006-07-07
申请人： Yoshiaki Takagi , Kok Seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono
发明人： Yoshiaki Takagi , Kok Seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono
IPC分类号： H03G3/00
CPC分类号： G10L19/008 , G10L2021/02082
摘要： Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device includes: a decorrelated signal generation unit, and a matrix operation unit and a third arithmetic unit. The decorrelated signal generation unit generates a decorrelated signal w′ indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit and the third arithmetic unit generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w′ generated by the decorrelated signal generation unit, using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.
摘要翻译：提供了一种减少算术运算负载的多声道声信号处理装置。多声道声信号处理装置包括：去相关信号生成单元，矩阵运算单元和第三运算单元。去相关信号生成单元通过对输入信号x进行混响处理，生成表示包含由输入信号x表示的声音和混响的声音的解相关信号w'。矩阵运算单元和第三运算单元通过使用表示信号强度分布的矩阵R3对输入信号x和解相关信号生成单元生成的去相关信号w'进行算术运算，生成m声道的音频信号水平和分布的混响。

10. 发明申请

US20090262949A1 MULTI-CHANNEL ACOUSTIC SIGNAL PROCESSING DEVICE 有权
标题翻译：多通道声信号处理装置
公开(公告)号：US20090262949A1
公开(公告)日：2009-10-22
申请号：US12064975
申请日：2006-07-07
申请人： Yoshiaki Takagi , Kok Seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono
发明人： Yoshiaki Takagi , Kok Seng Chong , Takeshi Norimatsu , Shuji Miyasaka , Akihisa Kawamura , Kojiro Ono
IPC分类号： H03G3/00
CPC分类号： G10L19/008 , G10L2021/02082
摘要： Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device (100) includes: a decorrelated signal generation unit (181), and a matrix operation unit (187) and a third arithmetic unit (186). The decorrelated signal generation unit (181) generates a decorrelated signal w′ indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit (187) and the third arithmetic unit (186) generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w′ generated by the decorrelated signal generation unit (181), using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.
摘要翻译：提供了一种减少算术运算负载的多声道声信号处理装置。多通道声信号处理装置（100）包括：解相关信号生成部（181），矩阵运算部（187）和第三运算部（186）。解相关信号生成单元（181）通过对输入信号x进行混响处理，生成表示包含由输入信号x表示的声音和混响的声音的解相关信号w'。矩阵运算单元（187）和第三运算单元（186）通过对输入信号x进行算术运算和由解相关信号生成单元（181）生成的去相关信号w'，使用矩阵R3，其指示信号强度水平的分布和混响的分布。

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