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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
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
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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
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。
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公开(公告)号：US5684881A
公开(公告)日：1997-11-04
申请号：US247269
申请日：1994-05-23
申请人： Mitsuhiko Serikawa , Ryou Tagami , Akihisa Kawamura , Masaharu Matsumoto , Mikio Oda , Hiroko Numazu
发明人： Mitsuhiko Serikawa , Ryou Tagami , Akihisa Kawamura , Masaharu Matsumoto , Mikio Oda , Hiroko Numazu
摘要： The apparatus of the invention calculates filter coefficients for controlling sound field and sound image, based on a plurality of first impulse response signals and a pair of second impulse response signals. The plurality of first impulse response signals indicate impulse responses from loudspeakers reproducing audio signals to both ears of a listener. The pair of second impulse response signals indicate impulse responses from a reference loudspeaker at a position at which a sound image is localized to both ears of the listener. The apparatus includes: a feature extracting section for receiving the pair of second impulse response signals, for extracting parameters representing features of the pair of second impulse response signals, and for outputting parameter signals; a signal adjusting section for adjusting at least one of the plurality of first impulse response signals based on the parameter signals, and for outputting a pair of third impulse response signals having the same features as the extracted features; and a coefficient calculating section for calculating the filter coefficients for controlling the sound field and sound image, based on the plurality of first impulse response signals and the pair of third impulse response signals applied from the signal adjusting means.
摘要翻译： 本发明的装置基于多个第一脉冲响应信号和一对第二脉冲响应信号来计算用于控制声场和声像的滤波器系数。 多个第一脉冲响应信号指示从再现音频信号的扬声器到听众的双耳的脉冲响应。 一对第二脉冲响应信号指示在声音图像被定位在收听者的两只耳朵的位置处的参考扬声器的脉冲响应。 该装置包括：特征提取部分，用于接收一对第二脉冲响应信号，用于提取表示该对第二脉冲响应信号的特征的参数，并用于输出参数信号; 信号调整部分，用于基于参数信号调整多个第一脉冲响应信号中的至少一个，并输出具有与提取的特征相同特征的一对第三脉冲响应信号; 以及系数计算部分，用于基于从信号调节装置施加的多个第一脉冲响应信号和一对第三脉冲响应信号，来计算用于控制声场和声像的滤波器系数。
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公开(公告)号：US5796845A
公开(公告)日：1998-08-18
申请号：US883276
申请日：1997-06-26
申请人： Mitsuhiko Serikawa , Ryou Tagami , Akihisa Kawamura , Masaharu Matsumoto , Mikio Oda , Hiroko Numazu
发明人： Mitsuhiko Serikawa , Ryou Tagami , Akihisa Kawamura , Masaharu Matsumoto , Mikio Oda , Hiroko Numazu
摘要： The apparatus of the invention calculates filter coefficients for controlling sound field and sound image, based on a plurality of first impulse response signals and a pair of second impulse response signals. The plurality of first impulse response signals indicate impulse responses from loudspeakers reproducing audio signals to both ears of a listener. The pair of second impulse response signals indicate impulse responses from a reference loudspeaker at a position at which a sound image is localized to both ears of the listener. The apparatus includes: a feature extracting section for receiving the pair of second impulse response signals, for extracting parameters representing features of the pair of second impulse response signals, and for outputting parameter signals; a signal adjusting section for adjusting at least one of the plurality of first impulse response signals based on the parameter signals, and for outputting a pair of third impulse response signals having the same features as the extracted features; and a coefficient calculating section for calculating the filter coefficients for controlling the sound field and sound image, based on the plurality of first impulse response signals and the pair of third impulse response signals applied from the signal adjusting means.
摘要翻译： 本发明的装置基于多个第一脉冲响应信号和一对第二脉冲响应信号来计算用于控制声场和声像的滤波器系数。 多个第一脉冲响应信号指示从再现音频信号的扬声器到听众的双耳的脉冲响应。 一对第二脉冲响应信号指示在声音图像被定位在收听者的两只耳朵的位置处的参考扬声器的脉冲响应。 该装置包括：特征提取部分，用于接收一对第二脉冲响应信号，用于提取表示该对第二脉冲响应信号的特征的参数，并用于输出参数信号; 信号调整部分，用于基于参数信号调整多个第一脉冲响应信号中的至少一个，并输出具有与提取的特征相同特征的一对第三脉冲响应信号; 以及系数计算部分，用于基于从信号调节装置施加的多个第一脉冲响应信号和一对第三脉冲响应信号，来计算用于控制声场和声像的滤波器系数。
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公开(公告)号：US5442712A
公开(公告)日：1995-08-15
申请号：US113658
申请日：1993-08-31
CPC分类号： H04R3/02
摘要： A sound signal picked-up by a microphone is processed in a howl suppresser including a digital filter. A frequency analyzer performs frequency analysis of the picked-up sound signal. A howl detector detects a howl contained in the sound signal from a result of frequency analysis by the frequency analyzer. A coefficient calculator calculates coefficients to be input to the digital filter to suppress the howl according to a detection result by the howl detector, and a controller inputs the calculated coefficients to the digital filter. The howl detector judges that a maximum peak power level among power levels of the sound signal in a frequency region analyzed by the frequency analyzer is a howl component when a ratio of the maximum peak power level to a mean power level of the sound signal is larger than a predetermined threshold level, preferably for a predetermined threshold time. Also, a threshold controller is provided for controlling the threshold level and/or the threshold time so as to enhance the accuracy of the howl detection.
摘要翻译： 在包括数字滤波器的嚎叫抑制器中处理由麦克风拾取的声音信号。 频率分析仪对拾取的声音信号进行频率分析。 嚎叫检测器通过频率分析仪从频率分析的结果中检测包含在声音信号中的嚎叫。 系数计算器计算要输入到数字滤波器的系数，以根据嚎叫检测器的检测结果抑制嚎叫，并且控制器将所计算的系数输入到数字滤波器。 嚎叫检测器判断当频率分析器分析的频率区域中的声音信号的功率电平之间的最大峰值功率电平是当最大峰值功率电平与声音信号的平均功率电平的比较大时的嚎叫分量 优选地达到预定阈值水平，优选地达到预定阈值时间。 此外，提供阈值控制器用于控制阈值水平和/或阈值时间，以便提高嚎叫检测的准确度。
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公开(公告)号：US5375188A
公开(公告)日：1994-12-20
申请号：US896044
申请日：1992-06-08
CPC分类号： G10L25/78 , G10H1/0091 , H04S7/30 , G10H2210/046 , H04S7/305
摘要： A music/voice discriminating apparatus is composed of a signal processing portion for effecting the signal processing upon input acoustic signals, a music/voice deciding portion for discriminating whether the input acoustic signals are music or voice, a first signal processing portion for optimally setting acoustic parameters for the signal processing respectively for music or voice, and a second signal processing portion for controlling the acoustic parameters of the first signal processing portion in accordance with the decision results of the music/voice deciding portion so that it may become a desirable value set in the second parameter setting portion.
摘要翻译： 音乐/语音识别装置由用于对输入的声音信号进行信号处理的信号处理部分，用于识别输入的声音信号是音乐还是语音的音乐/声音判定部分，用于最佳地设置声音的第一信号处理部分 用于分别用于音乐或语音的信号处理的参数，以及第二信号处理部分，用于根据音乐/声音决定部分的判定结果来控制第一信号处理部分的声学参数，使得它可以成为期望的值集合 在第二参数设置部分中。
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公开(公告)号：US5233661A
公开(公告)日：1993-08-03
申请号：US687695
申请日：1991-04-19
摘要： A sound field is generated responsive to direct sound picked up by a direct-sound collecting microphone and indirect-sound picked up by an indirect-sound collecting microphone. Respective processing circuits are provided for signal processing the sound signals of the direct-sound collecting microphone and the indirect-sound collecting microphone. An adaptive filter is provided having a transfer function equal to a transfer function from a reproducing speaker to the indirect-sound collecting microphone. The output of the signal processing circuit associated with the indirect-sound collecting microphone is filtered by the adaptive filter, and the output of the adaptive filter is subtracted from the output of the indirect-sound collecting microphone. The output of the subtractor is then applied to the signal processing circuit associated with the indirect-sound collecting microphone for signal processing. The transfer function is initially set in the adaptive filter by applying an output of a signal generator thereto.
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8.发明授权
公开(公告)号：US5164840A
公开(公告)日：1992-11-17
申请号：US400190
申请日：1989-08-28
申请人： Akihisa Kawamura , Masaharu Matsumoto , Mitsuhiko Serikawa , Ryo Tagami , Katsuaki Sato , Hiroko Yoshida , Hikari Hashimoto
发明人： Akihisa Kawamura , Masaharu Matsumoto , Mitsuhiko Serikawa , Ryo Tagami , Katsuaki Sato , Hiroko Yoshida , Hikari Hashimoto
CPC分类号： G11B20/00992 , G11B27/007 , G11B27/107 , G11B27/322 , G11B27/34 , H04N9/802 , H04S1/007
摘要： Control codes for controlling processing of an audio signal by a sound field variation circuit to produce desired sound fields are recorded on a monaural track of a video tape concurrent with recording of video and audio signals, and can thereby be subsequently played back for producing sound fields which are respectively synchronized with video images.
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公开(公告)号：US5144673A
公开(公告)日：1992-09-01
申请号：US623619
申请日：1990-12-07
摘要： A reflection sound compression apparatus extracts and compresses, most appropriately with a physical evaluation value, an impulse response of a hall, etc. which is obtained by calculation and actual experiments to reflection sounds in a number required by a sound field controller by using a learning identification method.
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公开(公告)号：US5073942A
公开(公告)日：1991-12-17
申请号：US645476
申请日：1991-01-24
CPC分类号： G10H1/0091 , G10H2210/301 , Y10S84/26
摘要： A sound field switcher inputs an output sound field signal from a sound field producer. The inputted sound field signal is delayed sequentially by plural delayers, after which they are multiplied by coefficients by plural multipliers, and the multiplied results are all added by an adder. Coefficients stored in an area of a memory indicated by a coefficient counter are read out and set to the plural multipliers. The sound field switcher carries out control of the timing for switching the sound field signal produced by the sound field producer and the timing for the coefficient counter to commence counting. Assuming the time required for switching the sound field to be N sampling periods, in case of carrying out a sound field switching from a sound field A to a sound field B, a sound field switching controller issues an order to the coefficient counter so as to count up one by one from 0 to N, and on each count the coefficients stored in an area of the coefficient memory is set to the multipliers. When the count value of the coefficient counter comes to (N/2) or nearest to (N/2), the sound field switching controller issues an order to the sound field producer to output the sound field signal of the sound field B. When the count value of the coefficient counter comes to N, the switching from the sound field A to the sound field B is completed.
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