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1. 发明授权

US06665645B1 Speech recognition apparatus for AV equipment 有权
标题翻译： AV设备语音识别装置
公开(公告)号：US06665645B1
公开(公告)日：2003-12-16
申请号：US09627071
申请日：2000-07-27
申请人： Satoru Ibaraki , Takeo Kanamori , Takashi Kawamura , Mikio Oda
发明人： Satoru Ibaraki , Takeo Kanamori , Takashi Kawamura , Mikio Oda
IPC分类号： G10L1102
CPC分类号： G10L15/20 , G10L2021/02082 , G10L2021/02161
摘要： A low-cost speech recognition apparatus for AV equipment capable of speech recognition with high accuracy while 2-channel sound is being produced from loudspeakers is achieved. A monaural conversion part converts 2-channel signals to be inputted to the loudspeakers into a monaural signal. A single echo canceller is provided with an output from a microphone and an output from the monaural conversion part (monaural signal). The echo canceller estimates an echo of multichannel sound based on the monaural signal, and then eliminates the echo sound from the microphone output. Thus, with only a single echo canceller, speech recognition can be carried out while 2-channel sound is being produced from the loudspeakers. Moreover, unlike the case where two echo cancellers are provided, the present invention can prevent the occurrence of mutual interference between the echo cancellers that leads to deterioration in speech recognition performance.
摘要翻译：实现了一种用于AV设备的低成本语音识别装置，其能够在从扬声器产生2声道声音的同时以高精度进行语音识别。单声道转换部分将要输入到扬声器的2声道信号转换为单声道信号。单个回声消除器具有来自麦克风的输出和来自单声道转换部分（单声道信号）的输出。回声消除器基于单声道信号估计多声道声音的回波，然后消除来自麦克风输出的回声。因此，仅使用单个回声消除器，可以在从扬声器产生2声道声音的同时进行语音识别。此外，与提供两个回波消除器的情况不同，本发明可以防止导致语音识别性能劣化的回波消除器之间的相互干扰的发生。

2. 发明授权

US06639986B2 Built-in microphone device 有权
标题翻译：内置麦克风设备
公开(公告)号：US06639986B2
公开(公告)日：2003-10-28
申请号：US09334493
申请日：1999-06-16
申请人： Takeo Kanamori , Takashi Kawamura , Satoru Ibaraki
发明人： Takeo Kanamori , Takashi Kawamura , Satoru Ibaraki
IPC分类号： H04B1500
CPC分类号： H04R3/005
摘要： A built-in microphone device for use in an apparatus having a mechanism section generating internal noise inside a housing of the apparatus includes a main microphone for picking up an external sound; a noise reference microphone for picking up the internal noise; an adaptive filter member for generating a control audio signal based on an output signal from the noise reference microphone using a filter coefficient; a signal subtraction section for subtracting the control audio signal generated by the adaptive filter member from an output signal from the main microphone to generate a subtraction result; and a filter coefficient update control section for receiving an operation signal generated at the time of an operation of the mechanism section, and in response to the operation signal, updating the filter coefficient of the adaptive filter member based on the subtraction result generated by the signal subtraction section and an output signal from the noise reference microphone.
摘要翻译：一种内置的麦克风装置，用于具有在该装置的壳体内产生内部噪声的机构部分的装置，包括用于拾取外部声音的主麦克风; 用于拾取内部噪声的噪声参考麦克风; 自适应滤波器构件，用于使用滤波器系数基于来自噪声参考麦克风的输出信号生成控制音频信号; 信号减法部分，用于从主麦克风的输出信号中减去由自适应滤波器部件产生的控制音频信号以产生减法结果; 以及滤波器系数更新控制部分，用于接收在机构部分的操作时产生的操作信号，并且响应于操作信号，基于由信号产生的减法结果来更新自适应滤波器部件的滤波器系数减法部分和来自噪声参考麦克风的输出信号。

3. 发明授权

US5058170A Array microphone 失效
标题翻译：阵列麦克风
公开(公告)号：US5058170A
公开(公告)日：1991-10-15
申请号：US473398
申请日：1990-02-01
申请人： Takeo Kanamori , Hiroki Furukawa , Satoru Ibaraki , Michio Matsumoto
发明人： Takeo Kanamori , Hiroki Furukawa , Satoru Ibaraki , Michio Matsumoto
IPC分类号： G10K11/34 , H04R1/40 , H04R3/00 , H04R3/02
CPC分类号： H04R3/005 , G10K11/341 , H04R1/406 , H04R2201/403 , H04R2201/405
摘要： An array microphone is a directional microphone having an improved directional characteric in which both the sensitivity and the sound pressure frequency response are uniform within the recording area and more particularly, the quality and level of sound remain unchanged. The array microphone includes a microphone array including a plurality of microphone units, and a two-dimensional filter for filtering an output of the microphone array in the dimensions of both time and space. When the two-dimensional filter is a digital filter and varied in its two-dimensional filter coefficient and sampling frequency, the array microphone serves as a variable directional microphone whose directional characteristic can be varied with the sound quality and level remaining unchanged throughout the recording area.

4. 发明授权

US5633935A Stereo ultradirectional microphone apparatus 失效
标题翻译：立体声超方向麦克风设备
公开(公告)号：US5633935A
公开(公告)日：1997-05-27
申请号：US225625
申请日：1994-04-11
申请人： Takeo Kanamori , Junichi Tagawa , Satoru Ibaraki , Hiroki Furukawa , Kiminori Ono
发明人： Takeo Kanamori , Junichi Tagawa , Satoru Ibaraki , Hiroki Furukawa , Kiminori Ono
IPC分类号： H04R3/00 , H04S1/00 , H04R5/00
CPC分类号： H04R3/005 , H04S1/002
摘要： A stereo ultradirectional microphone apparatus for detecting a sound to produce stereo sound signals, comprises: first and second ultradirectional microphones arranged side by side with a given distance in parallel for converting a sound into first and second sound signals respectively, first and second delays for delaying an output of the first and second microphones by a delay time .tau. respectively, and first and second subtractors for subtracting for subtraction between the first sound signal and an output of the second delay and subtracting for subtraction between the second sound signal and an output of the first delay. The delay time .tau. corresponds to a difference between the timings of a sound from a sound source in a direction making a clockwise angel .theta. from the front where a dead angel should be made. The subtraction provides the dead angle. Similarly, a dead angle on the side is also made to obtain a stereo characteristic. The directivity in the frequency characteristic of microphone is equalized to cancel the sensitivity in the dead angle. A stereo apparatus for forming the dead angles with two set of two filters having respective transfer characteristics determined by measurement and an adder is also disclosed.
摘要翻译：一种用于检测声音以产生立体声信号的立体声超方向麦克风装置包括：并排布置给定距离的第一和第二超向量麦克风，用于将声音分别转换成第一和第二声音信号，用于延迟的第一和第二延迟第一和第二麦克风的输出分别为延迟时间τ，以及第一和第二减法器，用于减去第一声音信号和第二延迟的输出之间的减法，并减去第二声音信号与第二声音信号的输出之间的减法第一次延迟延迟时间ττ对应于来自声源的声音的定时之间的差异，该方向从应该制造死角的前方形成顺时针方位角θ的方向。减法提供死角。类似地，还制造了一侧的死角以获得立体声特性。麦克风频率特性的方向性被均衡以抵消死角的灵敏度。还公开了一种用于通过由测量确定的具有各自的传送特性的两组滤波器和加法器形成死角的立体声装置。

5. 发明授权

US5463618A Echo canceller 失效
标题翻译：回音消除器
公开(公告)号：US5463618A
公开(公告)日：1995-10-31
申请号：US249888
申请日：1994-05-26
申请人： Hiroki Furukawa , Junichi Tagawa , Takeo Kanamori , Satoru Ibaraki
发明人： Hiroki Furukawa , Junichi Tagawa , Takeo Kanamori , Satoru Ibaraki
IPC分类号： H04B3/23 , H04M9/08
CPC分类号： H04B3/234 , H04M9/082
摘要： In an echo canceller, a first adaptive filter (1) estimates an impulse response of an echo path between reception and transmission side by an NLMS algorithm to synthesize a pseud echo, while a second adaptive filter (3) determines whether there is an echo path change or a double talk. A voice detector (8) detects a short time power of the received input signal (x) to determine whether far-end speech is present or absent. The voice detector controls the second adaptive filter (3) to execute adaptation of the second adaptive filter for renewal of an impulse response of an echo path when far-end speech is present and otherwise to suspend the adaptation of the second adaptive filter when far-end speech is absent, while a double talk detector (9) controls the first adaptive filter to execute or suspend adaptation of the first adaptive filter according to the determination results of the voice detector.
摘要翻译：在回波消除器中，第一自适应滤波器（1）通过NLMS算法估计接收侧和发送侧之间的回波路径的脉冲响应以合成伪回波，而第二自适应滤波器（3）确定是否存在回波路径改变或双重谈话。语音检测器（8）检测接收的输入信号（x）的短时间功率，以确定是否存在远端语音。语音检测器控制第二自适应滤波器（3）执行第二自适应滤波器的适应，以便在存在远端语音时对回波路径的脉冲响应进行更新，并且当远端语音存在时暂停第二自适应滤波器的自适应，结束语言不存在，而双通话检测器（9）根据语音检测器的确定结果控制第一自适应滤波器来执行或暂停适应第一自适应滤波器。

6. 发明授权

US07577262B2 Microphone device and audio player 有权
标题翻译：麦克风设备和音频播放器
公开(公告)号：US07577262B2
公开(公告)日：2009-08-18
申请号：US10714857
申请日：2003-11-18
申请人： Takeo Kanamori , Takashi Kawamura , Tomomi Matsuoka
发明人： Takeo Kanamori , Takashi Kawamura , Tomomi Matsuoka
IPC分类号： H04B15/00
CPC分类号： H04R3/005 , H04R2410/01 , H04R2410/05
摘要： A signal generating section generates a main signal and a noise reference signal. A determining section determines whether a level ratio is larger than a predetermined value. An adaptive filter section generates a signal indicative of a signal component of a target sound included in the noise reference signal generated by the signal generating section, and learns a filter coefficient only when the determining section determines that the level ratio is larger than the predetermined value. A subtracting section subtracts the signal generated by the adaptive filter section from the noise reference signal. A noise suppressing section suppresses a signal component of noise included in the main signal by using the main signal and the noise reference signal after subtraction by the subtracting section.
摘要翻译：信号产生部分产生主信号和噪声参考信号。确定部分确定电平比是否大于预定值。自适应滤波器部分产生指示由信号产生部分生成的噪声参考信号中包括的目标声音的信号分量的信号，并且仅在确定部分确定电平比大于预定值时才学习滤波器系数。减法部从噪声参考信号中减去由自适应滤波器部分产生的信号。噪声抑制部分通过减法部分减去后的主信号和噪声参考信号来抑制包含在主信号中的噪声的信号分量。

7. 发明授权

US07567900B2 Harmonic structure based acoustic speech interval detection method and device 有权
标题翻译：基于谐波结构的声学语音间隔检测方法及装置
公开(公告)号：US07567900B2
公开(公告)日：2009-07-28
申请号：US10542931
申请日：2004-06-03
申请人： Tetsu Suzuki , Takeo Kanamori , Takashi Kawamura
发明人： Tetsu Suzuki , Takeo Kanamori , Takashi Kawamura
IPC分类号： G10L15/20 , G10L11/06 , G10L15/00 , G10L17/00
CPC分类号： G10L25/78 , G10L2025/932 , G10L2025/937
摘要： A harmonic structure acoustic signal detection device not depending on the level fluctuation of the input signal including: an FFT unit which performs FFT on an input signal and calculates a power spectrum component for each frame; a harmonic structure extraction unit which leaves only a harmonic structure from the power spectrum component; a voiced feature evaluation unit which evaluates correlation between the frames of harmonic structures extracted by the harmonic structure extraction unit, thereby evaluates whether or not the segment is a vowel segment, and extracts the voiced segment; and a speech segment determination unit which determines a speech segment according to the continuity and durability of the output of the voiced feature evaluation unit.
摘要翻译：一种谐波结构声信号检测装置，其不依赖于输入信号的电平波动，包括：FFT单元，对输入信号执行FFT并计算每帧的功率谱分量; 谐波结构提取单元，仅从功率谱分量中留下谐波结构; 声音特征评估单元，其评估由所述谐波结构提取单元提取的谐波结构的相关性，从而评估所述段是否是元音段，并且提取所述有声段; 以及语音段确定单元，其根据有声特征评估单元的输出的连续性和耐久性来确定语音段。

8. 发明申请

US20080021703A1 Howling Detection Device and Method 失效
标题翻译：啸叫侦测装置及方法
公开(公告)号：US20080021703A1
公开(公告)日：2008-01-24
申请号：US11579133
申请日：2005-06-15
申请人： Takashi Kawamura , Takeo Kanamori
发明人： Takashi Kawamura , Takeo Kanamori
IPC分类号： H04R3/02
CPC分类号： H04R3/005 , H04R3/02
摘要： A howling detection device detects a dominance ratio, which indicates a risk of howling to be occurred when a mixed signal obtained by a sound mixing section for mixing a plurality of sound signals respectively collected by a plurality of microphones is outputted by a speaker, for each of the sound signals. The howling detection device includes a level detecting section for respectively detecting levels of the plurality of sound signals, a word ending detecting section for comparing, in a same time domain, the mixed signal with a signal regarding a sound to be outputted by the speaker as a noise reference signal, and detecting a time period, as a word ending section, during which the mixed signal is inputted after the noise reference signal falls, and a dominance ratio calculating section for extracting only any of the levels corresponding to the word ending section from among the levels, of the plurality of sound signals, detected by the level detecting section, and calculating, as a dominance ratio, a ratio of each of the levels of each of the sound signals to a sum of the levels of the plurality of sound signals.
摘要翻译：吵音检测装置检测优势比，其表示当由扬声器输出由扬声器输出由用于混合由多个麦克风分别收集的多个声音信号的声音混合部分获得的混合信号时发生啸声的风险的声音信号。啸声检测装置包括用于分别检测多个声音信号的电平的电平检测部分，一个字尾结束检测部分，用于将混合信号与扬声器输出的声音相关的信号在同一时域中进行比较噪声参考信号，并且检测在噪声参考信号下降之后输入混合信号的时间段作为字结束部分，以及优势比计算部分，用于仅提取与字结束部分相对应的任何电平从电平检测部检测到的多个声音信号的电平之中，计算出每个声音信号的每个电平与多个声音信号的电平之和作为优势比，声音信号。

9. 发明授权

US06931138B2 Zoom microphone device 有权
标题翻译：变焦麦克风设备
公开(公告)号：US06931138B2
公开(公告)日：2005-08-16
申请号：US09983284
申请日：2001-10-23
申请人： Takashi Kawamura , Takeo Kanamori , Toshihide Ishimoto
发明人： Takashi Kawamura , Takeo Kanamori , Toshihide Ishimoto
IPC分类号： H04R1/40 , H04R3/00 , H04N5/76
CPC分类号： H04R3/005 , H04R1/406 , H04R2201/401 , H04R2201/403
摘要： A pickup section transduces sounds to audio signals. A zoom control section outputs a zoom position signal corresponding to a zoom position. A directivity control section alters the directivity characteristics under a telescopic operation so as to mainly pick up sounds coming from a frontal direction with an enhancement which is in accordance with the zoom position signal, thereby outputting an R channel audio signal and an L channel audio signal. In accordance with the zoom position signal, a noise suppression section applies a greater degree of suppression to the background noise contained in the respective channel audio signals under the telescopic operation than under a wide-angle operation. As a result, a target sound from a remote location can be picked up with a sufficient enhancement in accordance with the zoom position under the telescopic operation.
摘要翻译：拾音部分将声音转换为音频信号。变焦控制部输出与变焦位置对应的变焦位置信号。指向性控制部分在伸缩操作下改变方向性特征，主要从正面方向拾取声音，并且具有与变焦位置信号相应的增强，从而输出R声道音频信号和L声道音频信号。根据变焦位置信号，噪声抑制部分对于在伸缩操作下的相应声道音频信号中包含的背景噪声比在广角操作下更大程度的抑制。结果，可以根据伸缩操作下的变焦位置以足够的增强来拾取来自远程位置的目标声音。

10. 发明授权

US07912230B2 Howling detection device and method 失效
标题翻译：啸叫检测装置及方法
公开(公告)号：US07912230B2
公开(公告)日：2011-03-22
申请号：US11579133
申请日：2005-06-15
申请人： Takashi Kawamura , Takeo Kanamori
发明人： Takashi Kawamura , Takeo Kanamori
IPC分类号： H04B15/00 , H04R27/00
CPC分类号： H04R3/005 , H04R3/02
摘要： A howling detection device detects a dominance ratio, which indicates a risk of howling to occur when a mixed signal obtained by mixing a plurality of sound signals collected by a plurality of microphones is outputted by a speaker. The howling detection device detects levels of the plurality of sound signals, compares, in a same time domain, the mixed signal with a signal regarding a sound to be outputted by the speaker as a noise reference signal, detects a time period, as a word ending section, during which the mixed signal is inputted after the noise reference signal falls, and calculates a dominance ratio by extracting only a level of the plurality of sound signals corresponding to the word ending section and determining a ratio of each of the extracted levels of each of the sound signals to a sum of the extracted levels of the plurality of sound signals.
摘要翻译：当通过由扬声器输出通过混合由多个麦克风采集的多个声音信号而获得的混合信号时，啸叫检测装置检测优势比，其表示啸叫发生的风险。啸声检测装置检测多个声音信号的电平，将混合信号与由扬声器输出的声音相关的信号作为噪声参考信号进行比较，将时间段作为一个字结束部分，其中混合信号在噪声参考信号下降之后被输入，并且通过仅提取对应于字结束部分的多个声音信号的电平来计算优势比，并且确定每个提取电平的比率声音信号中的每一个与所提取的多个声音信号的电平之和。

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