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1. 发明申请

US20110157797A1 CASING, ELECTRONIC APPARATUS, AND METHOD AND MACHINE FOR FORMING CASING 有权
标题翻译：铸造，电子设备，以及用于成型铸造的方法和机器
公开(公告)号：US20110157797A1
公开(公告)日：2011-06-30
申请号：US12975940
申请日：2010-12-22
申请人： Toru Kurihara , Masayuki Sato , Yutaka Kawahigashi , Kazuhiro Kakuguchi , Nozomu Saito
发明人： Toru Kurihara , Masayuki Sato , Yutaka Kawahigashi , Kazuhiro Kakuguchi , Nozomu Saito
IPC分类号： H05K5/03 , B65D25/00 , B29C45/00
CPC分类号： B29C45/0081 , B29C45/006 , B29C45/33 , B29C45/44 , B29C45/4471 , B29L2031/3437
摘要： A casing includes a first casing and a second casing that sandwich an elastic member, wherein the first casing includes a pressed portion that is pressed by the elastic member, and an engaging portion adjacent to the pressed portion that has a flange protruding toward the pressed portion that engages with the second casing, and the pressed portion is flat in a region corresponding to at least the engaging portion.
摘要翻译：壳体包括夹着弹性构件的第一壳体和第二壳体，其中，所述第一壳体包括被所述弹性构件按压的按压部，以及与所述按压部相邻的接合部，所述接合部具有朝向所述按压部突出的凸缘所述第二壳体与所述第二壳体接合，并且所述按压部分在至少所述接合部分对应的区域中是平坦的。

2. 发明授权

US07254242B2 Acoustic signal processing apparatus and method, and audio device 失效
标题翻译：声信号处理装置及方法及音响装置
公开(公告)号：US07254242B2
公开(公告)日：2007-08-07
申请号：US10453918
申请日：2003-06-03
申请人： Tomohiko Ise , Nozomu Saito
发明人： Tomohiko Ise , Nozomu Saito
IPC分类号： H04B15/00
CPC分类号： G10L21/02 , G10L2021/02082
摘要： A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.
摘要翻译：第一频带分析仪将通过输入单元从声音播放系统接收的声学信号划分成频带，并产生第一频带电平。声信号估计器估计输入单元处的原始声信号的频带电平，并为每个频带产生第二频带电平。处理器使用第一频带电平和第二频带电平提取包含在声学信号中的外部噪声分量。可以比现有技术更少的计算来精确地估计外部噪声。

3. 发明授权

US06778601B2 Adaptive audio equalizer apparatus and method of determining filter coefficient 失效
标题翻译：自适应音频均衡器装置及其滤波器系数的确定方法
公开(公告)号：US06778601B2
公开(公告)日：2004-08-17
申请号：US09777370
申请日：2001-02-05
申请人： Tomohiko Ise , Nozomu Saito
发明人： Tomohiko Ise , Nozomu Saito
IPC分类号： H04B110
CPC分类号： H03H21/0012
摘要： In an adaptive audio equalizer apparatus, a signal that is output from an adaptive filter 13 is fed to a speaker and to a delaying unit. The signal fed to the delaying unit is delayed for a predetermined time, and is multiplied by a scaling factor in a multiplying unit. A first calculation unit calculates a difference between an output of a microphone and a target response signal, and outputs the result as an error signal. A second calculation unit adds the output of the multiplying unit to the error signal, and outputs the result to a filter coefficient setting unit of the adaptive filter.
摘要翻译：在自适应音频均衡器装置中，从自适应滤波器13输出的信号被馈送到扬声器和延迟单元。馈送到延迟单元的信号延迟预定时间，并且乘以乘法单元中的缩放因子。第一计算单元计算麦克风的输出与目标响应信号之间的差，并输出该结果作为误差信号。第二计算单元将乘法单元的输出与误差信号相加，并将结果输出到自适应滤波器的滤波器系数设置单元。

4. 发明授权

US08249259B2 Voice intelligibility enhancement system and voice intelligibility enhancement method 有权
标题翻译：语音清晰度增强系统和语音清晰度增强方法
公开(公告)号：US08249259B2
公开(公告)日：2012-08-21
申请号：US12339360
申请日：2008-12-19
申请人： Toru Marumoto , Nozomu Saito
发明人： Toru Marumoto , Nozomu Saito
IPC分类号： H04R29/00
CPC分类号： H03G3/32
摘要： In a voice intelligibility enhancement system that controls a gain of a voice signal based on noise power and voice power of the voice signal generated by a voice signal generation unit, it is detected whether the voice power is equal to or greater than a predetermined level, noise power output when the voice power is less than the predetermined level is measured and stored, noise power to be output when the voice power exceeds the predetermined level is estimated to be the stored noise power, and gain of a voice signal is controlled on the basis of the voice power and the estimated noise power.
摘要翻译：在语音清晰度增强系统中，基于由语音信号生成单元产生的语音信号的噪声功率和语音功率来控制语音信号的增益，检测语音功率是否等于或大于预定电平，测量并存储当语音功率小于预定电平时的噪声功率输出，当语音功率超过预定电平时要输出的噪声功率被估计为存储的噪声功率，并且语音信号的增益被控制在语音功率的基础和估计的噪声功率。

5. 发明授权

US07486797B2 Audio correcting apparatus 有权
标题翻译：音频校正装置
公开(公告)号：US07486797B2
公开(公告)日：2009-02-03
申请号：US10862820
申请日：2004-06-07
申请人： Toru Marumoto , Nozomu Saito
发明人： Toru Marumoto , Nozomu Saito
IPC分类号： H03G3/20
CPC分类号： H04R29/003 , H03G9/005 , H03G9/025 , H04N5/60 , H04N21/42203 , H04N21/4221 , H04N21/42222 , H04N21/439 , H04N21/4852 , H04R2499/15
摘要： An audio correcting apparatus includes a speaker provided on a television apparatus, a microphone provided on a remote controller, an identifying unit which identifies an acoustic characteristic from the speaker to the microphone, and an acoustic characteristic setting unit having the acoustic characteristic. A signal obtained by allowing an audio signal input to the speaker to pass through the acoustic characteristic setting unit, and a signal representing ambient noise are input to an audio-correcting filter and a loudness-compensation-gain calculating unit. Based on both signals, the sound pressure level of sound output from the speaker is corrected so that the sound output from the speaker is clearly heard when reaching the user without being affected by the ambient noise.
摘要翻译：音频校正装置包括设置在电视机上的扬声器，设置在遥控器上的麦克风，识别从扬声器到麦克风的声学特性的识别单元，以及具有声学特性的声学特性设定单元。通过允许输入到扬声器的音频信号通过声学特性设定单元和表示环境噪声的信号而获得的信号被输入到音频校正滤波器和响度补偿增益计算单元。基于这两个信号，校正从扬声器输出的声音的声压级，使得当到达用户而不受环境噪声的影响时，从扬声器输出的声音被清楚地听到。

6. 发明申请

US20070019825A1 In-vehicle audio processing apparatus 审中-公开
标题翻译：车载音响处理装置
公开(公告)号：US20070019825A1
公开(公告)日：2007-01-25
申请号：US11477725
申请日：2006-06-28
申请人： Toru Marumoto , Shingo Kiuchi , Nozomu Saito
发明人： Toru Marumoto , Shingo Kiuchi , Nozomu Saito
IPC分类号： H04B15/00
CPC分类号： H04R5/02
摘要： On the basis of status information on a vehicle collected by a status information input interface from a navigation device, an ECU, and sensors, an S/N ratio estimating unit estimates, as an S/N ratio, the level of the ratio between the power of a component corresponding to audio-device output sound y(j) and that corresponding to noise sound n(j) contained in a microphone output signal. A transfer-function variation estimating unit estimates the level of a variation in a transfer function of an audio-device output audio signal transfer system. An adaptive characteristics controller controls a characteristic of a coefficient updating operation of a tap coefficient of an FIR filter performed by a coefficient updating unit of an adaptive filter, i.e., an adaptation (learning) characteristic of the adaptive filter, in response to the S/N ratio level and the level of the variation in the transfer function.
摘要翻译：基于来自导航装置，ECU和传感器的状态信息输入界面收集的车辆的状态信息，S / N比估计单元估计作为S / N比的比率的水平对应于音频设备输出声音y（j）的部件的功率和对应于麦克风输出信号中包含的噪声声音n（j）的功率。传递函数变化估计单元估计音频设备输出音频信号传送系统的传递函数的变化程度。自适应特性控制器控制由自适应滤波器的系数更新单元执行的FIR滤波器的抽头系数的系数更新操作的特性，即自适应滤波器的适应（学习）特性，响应于S / N比等级和传递函数的变化水平。

7. 发明授权

US06959277B2 Voice feature extraction device 有权
标题翻译：语音特征提取装置
公开(公告)号：US06959277B2
公开(公告)日：2005-10-25
申请号：US09891876
申请日：2001-06-26
申请人： Shingo Kiuchi , Toshiaki Asano , Nozomu Saito
发明人： Shingo Kiuchi , Toshiaki Asano , Nozomu Saito
IPC分类号： G10L15/00 , G10L15/02 , G10L15/20 , G10L15/28 , G10L21/02
CPC分类号： G10L15/20 , G10L15/02 , G10L21/0208
摘要： In a conventional device for extracting voice features accurately without being influenced by noises, such as a voice recognition device, usually an input voice signal is processed first by a noise reduction system having the tap length N, and the result is FFT-processed by L-points, and then the power spectrum vector is calculated; accordingly, a one time operation requires N multiplications and (N−1) summations. The voice feature extraction device according to the invention receives a voice signal including noises from a microphone, which is processed by a window function operation unit, and thereafter FFT-processed by an FFT operation unit by L-points. A power calculation unit calculates a power spectrum vector of the input voice signal. However, a noise reduction system determines in advance a filter coefficient of this system and processes the coefficient to calculate a noise reduction coefficient, and the power spectrum vector is processed by this noise reduction system. Thereby, the voice feature extraction device of the invention reduces the processing volume to 1/(4N−2) in comparison to the conventional device, lightens the processing load of the processing unit, and increases the processing speed.
摘要翻译：在诸如语音识别装置的语音识别装置的声音特征准确地提取语音特征的常规装置中，通常由具有抽头长度N的噪声降低系统首先处理输入语音信号，并且将结果由L 点，然后计算功率谱矢量; 因此，一次操作需要N次乘法和（N-1）个求和。根据本发明的语音特征提取装置接收由麦克风产生的噪声的声音信号，由麦克风进行处理，由窗函数运算单元进行处理，然后由FFT运算单元用L点进行FFT处理。功率计算单元计算输入语音信号的功率谱矢量。然而，噪声降低系统预先确定该系统的滤波器系数并处理该系数以计算降噪系数，并且由该降噪系统处理功率谱矢量。因此，本发明的语音特征提取装置与常规装置相比将处理量减少到1 /（4N-2），减轻了处理单元的处理负荷，并且提高了处理速度。

8. 发明申请

US20050080626A1 Voice output device and method 审中-公开
标题翻译：语音输出装置及方法
公开(公告)号：US20050080626A1
公开(公告)日：2005-04-14
申请号：US10925874
申请日：2004-08-24
申请人： Toru Marumoto , Nozomu Saito
发明人： Toru Marumoto , Nozomu Saito
IPC分类号： G10L15/00 , G10K15/00 , G10L13/00 , G10L13/02 , G10L13/08
CPC分类号： G10L13/02
摘要： Voice output device and method to generate voice messages that are highly comprehensible. The voice output device includes a voice database in which information indicating the familiarity level of each word or word string has been recorded, and a sound pressure adjustor for adjusting the sound pressure level of each word or word string on the basis of voice data and familiarity information read together with voice data from the voice database by a reproducer. For a word or the like having low familiarity, the sound pressure thereof is corrected by increasing it. Thus, to generate a voice message including a word of low familiarity, such as an unfamiliar place name, adjustment is performed so that the unfamiliar place name is generated with a higher sound pressure, as compared with a word of high familiarity. This allows words with low familiarity to be easily comprehended.
摘要翻译：语音输出设备和方法来产生高度易懂的语音留言。语音输出装置包括语音数据库，其中已经记录了表示每个单词或单词串的熟悉程度的信息，以及声压调节器，用于根据语音数据和熟悉度来调整每个单词或单词串的声压级通过再现器与语音数据库的语音数据一起读取信息。对于熟悉度低的单词等，通过增加声压来校正其声压。因此，为了生成包括诸如不熟悉的地名的低熟悉度的语音消息，与高熟悉的词相比，进行调整，使得不熟悉的地名以更高的声压产生。这样可以很容易理解熟悉程度低的词汇。

9. 发明授权

US07747028B2 Apparatus and method for improving voice clarity 有权
标题翻译：提高语音清晰度的装置和方法
公开(公告)号：US07747028B2
公开(公告)日：2010-06-29
申请号：US11070837
申请日：2005-03-02
申请人： Nozomu Saito , Toru Marumoto
发明人： Nozomu Saito , Toru Marumoto
IPC分类号： H03G3/00
CPC分类号： H03G3/32 , G10L21/0208 , G10L21/0264
摘要： In an apparatus for improving voice clarity by controlling a gain of a voice based on a sound pressure level of the voice and a sound pressure level of noise, it is determined whether a sound pressure level of a gain-controlled voice exceeds a maximum allowable level. If the sound pressure level exceeding the maximum allowable level is caused by audio sound, a sound pressure level of the audio sound is reduced. The sound pressure level of the audio sound is minimally reduced so that the sound pressure level of the voice becomes equal to or lower than the maximum allowable level.
摘要翻译：在通过基于声音的声压级和噪声的声压级控制声音的增益来改善语音清晰度的装置中，确定增益控制语音的声压级是否超过最大允许电平。如果超过最大允许电平的声压级由音频声音引起，音频声音的声压级就会降低。音频声音的声压级被最小化，使得声音的声压级别等于或低于最大允许电平。

10. 发明申请

US20090187406A1 VOICE RECOGNITION SYSTEM 失效
标题翻译：语音识别系统
公开(公告)号：US20090187406A1
公开(公告)日：2009-07-23
申请号：US12327209
申请日：2008-12-03
申请人： Kazunori Sakuma , Nozomu Saito , Tohru Masumoto
发明人： Kazunori Sakuma , Nozomu Saito , Tohru Masumoto
IPC分类号： G10L15/04 , G10L15/14
CPC分类号： G10L15/22 , G01C21/3608 , G10L2015/221
摘要： A voice recognition system is provided that outputs a talk-back voice in a manner such that a user can distinguish the accuracy of a voice-recognized character string more easily. A voice recognition unit performs voice recognition on a user's articulation in which a character string such as the telephone number “024 636 0123” is entered via a microphone. Based on each sound existing period delimited by silent intervals, each recognized partial character string “024”, “636” and “0123” is obtained. A talk-back voice data generating unit connects each recognized partial character string “024”, “636” and “0123” together in a manner such that space characters are inserted, and generates a character string “024 636 0123”. The generated character string “024 636 0123” is supplied to a voice generating device as talk-back voice data. A voice signal to be produced by the speaker 2 is generated in the form of the talk-back voice.
摘要翻译：提供了一种语音识别系统，其以使得用户可以更容易地区分语音识别字符串的精度的方式输出对讲语音。语音识别单元对用户的发音执行语音识别，其中通过麦克风输入诸如电话号码“024 636 0123”的字符串。基于由无声间隔分隔的每个声音存在期间，获得每个识别的部分字符串“024”，“636”和“0123”。对讲话音数据产生单元以插入空格字符的方式将每个识别的部分字符串“024”，“636”和“0123”连接在一起，并生成字符串“024 636 0123”。生成的字符串“024 636 0123”作为对话语音数据提供给语音生成装置。以讲话语音的形式产生由扬声器2产生的语音信号。

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