公开(公告)号：US08824700B2

公开(公告)日：2014-09-02

申请号：US13497299

申请日：2011-07-26

申请人： Takeo Kanamori ,  Shinichi Yuzuriha ,  Yutaka Banba ,  Yasuhiro Terada

发明人： Takeo Kanamori ,  Shinichi Yuzuriha ,  Yutaka Banba ,  Yasuhiro Terada

IPC分类号： H04B15/00 ,  G10L21/0208 ,  G10L21/0216

CPC分类号： G10L21/0208 ,  G10L2021/02165 ,  H04R2410/01 ,  H04R2410/05 ,  H04R2430/25

摘要： A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(ω), based on a power spectrum P1(ω) and on a first calculated value obtained by at least multiplying a power spectrum P2(ω) by a weight coefficient A2(ω). A coefficient update unit (300) updates the weight coefficient A2(ω) and a weight coefficient A1(ω) so that a second calculated value approximates to the power spectrum P1(ω). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(ω) and the estimated target sound power spectrum Ps(ω) by the weight coefficient A2(ω) and the weight coefficient A1(ω), respectively.

摘要翻译： 功率谱估计单元（200）基于功率谱P1（ω）获得估计声功率谱Ps（ω），并且通过至少将功率谱P2（ω）乘以权重系数获得的第一计算值 A2（ω）。 系数更新单元（300）更新权重系数A2（ω）和加权系数A1（ω），使得第二计算值近似于功率谱P1（ω）。 通过将功率谱P2（ω）和估计的目标声功率谱Ps（ω）乘以权重系数A2（ω）和权重系数A1（ω）而获得的至少两个值来获得第二计算值， 分别。

公开(公告)号：US20050143973A1

公开(公告)日：2005-06-30

申请号：US11041275

申请日：2005-01-25

申请人： Shohei Taniguchi ,  Masayuki Ito ,  Yutaka Banba

发明人： Shohei Taniguchi ,  Masayuki Ito ,  Yutaka Banba

IPC分类号： H03H17/02 ,  H04B1/66 ,  G10L19/00

CPC分类号： H03H17/0266 ,  H03H2021/0041

摘要： An apparatus having a band-separating filter bank for separating a digital signal into a plurality of sub-band signals, to be processed or transmitted, and a band-combining filter bank for subsequently combining the resultant sub-band signals into a single digital signal, wherein each of the band-separating filter bank and band-combining filter bank incorporates a FIR low pass filter having an asymmetric impulse response, as the prototype filter of the filter bank. A significant reduction can thereby be achieved in the amount of overall group delay that results from the processing performed by these filter banks.

摘要翻译： 一种具有用于将数字信号分离成待处理或发送的多个子带信号的带分离滤波器组的装置和用于随后将所得到的子带信号组合成单个数字信号的带合成滤波器组 其中，每个带分离滤波器组和频带组合滤波器组都包含具有非对称脉冲响应的FIR低通滤波器，作为滤波器组的原型滤波器。 因此，由这些滤波器组进行的处理所产生的整体组延迟的量可以实现显着的降低。

公开(公告)号：US06856653B1

公开(公告)日：2005-02-15

申请号：US09662823

申请日：2000-09-15

申请人： Shohei Taniguchi ,  Masayuki Ito ,  Yutaka Banba

发明人： Shohei Taniguchi ,  Masayuki Ito ,  Yutaka Banba

IPC分类号： H03H17/02 ,  H04B15/00

CPC分类号： H03H17/0266 ,  H03H2021/0041

摘要： An apparatus having a band-separating filter bank for separating a digital signal into a plurality of sub-band signals, to be processed or transmitted, and a band-combining filter bank for subsequently combining the resultant sub-band signals into a single digital signal, wherein each of the band-separating filter bank and band-combining filter bank incorporates a FIR low pass filter having an asymmetric impulse response, as the prototype filter of the filter bank. A significant reduction can thereby be achieved in the amount of overall group delay that results from the processing performed by these filter banks.

公开(公告)号：US07478309B2

公开(公告)日：2009-01-13

申请号：US10617100

申请日：2003-07-10

申请人： Yutaka Banba

发明人： Yutaka Banba

IPC分类号： H03M13/00

CPC分类号： H04L1/0059 ,  H04L1/0041 ,  H04L1/0061 ,  H04L1/007

摘要： An audio encoder converts an input sound signal into a plurality of compressed frame data pieces in an sound signal compression coder, determines the importance of each bit in a classification unit of a transmission line coder based on the decoding quality in the presence of a transmission error, and classifies the bits into a plurality of classes. The audio encoder selects one of the three types of processing including convolution coding and addition of CRC check codes, convolution coding only, and no coding, in descending order of importance in the presence of a transmission error for each class. Then, the audio encoder adds preamble information and a synchronization signal in a multiplexer to generate a bit stream. It becomes possible to suppress degradation of a decoded sound signal without additional redundant bits.

摘要翻译： 音频编码器将输入声音信号转换成声音信号压缩编码器中的多个压缩帧数据，在存在传输错误的情况下，基于解码质量来确定传输线编码器的分类单元中每个比特的重要性 ，并将这些位分类成多个类。 音频编码器在存在每个类的传输错误的情况下，按照重要性的降序，选择三种类型的处理，其中包括卷积编码和CRC校验码的加法，卷积编码，无编码。 然后，音频编码器在多路复用器中添加前同步码信息和同步信号以产生比特流。 可以抑制解码声音信号的劣化，而无需额外的冗余位。

公开(公告)号：US20130156208A1

公开(公告)日：2013-06-20

申请号：US13819349

申请日：2012-02-21

申请人： Yutaka Banba ,  Takeo Kanamori

发明人： Yutaka Banba ,  Takeo Kanamori

IPC分类号： H04R25/00

CPC分类号： H04R25/30 ,  G10L21/0208 ,  G10L2021/02165 ,  H04R25/407 ,  H04R2430/03

摘要： The present invention provides a hearing aid capable of detecting contact vibration noise from a collected sound signal. A hearing aid (100) is provided with two microphones (110-1, 110-2), a vibration component extracting unit (120) which extracts from collected sound signals respectively obtained by the two microphones (110-1, 110-2) an uncorrelated component between two collected sound signals as a vibration component for each frequency band, a vibration noise identifying unit (130) which determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting unit (120), an acoustic signal processing unit (140) which, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver (150) which converts the acoustic signal to sound.

摘要翻译： 本发明提供一种能够从收集的声音信号中检测接触振动噪声的助听器。 助听器（100）具有两个麦克风（110-1,110-2），从由两个麦克风（110-1,110-2）分别获得的收集的声音信号中提取的振动分量提取单元（120） 在两个收集的声音信号之间作为每个频带的振动分量的不相关分量，基于由振动分量提取单元提取的每个频带的振动分量来确定是否发生接触噪声的振动噪声识别单元（130） （120），声信号处理单元（140），当通过对所收集的两个声音信号进行助听处理产生声信号时，根据是否存在接触振动噪声来处理声信号，以及 接收器（150），其将声信号转换成声音。

公开(公告)号：US07155384B2

公开(公告)日：2006-12-26

申请号：US10277827

申请日：2002-10-23

申请人： Yutaka Banba

发明人： Yutaka Banba

IPC分类号： G10L19/02

CPC分类号： G10L19/10 ,  G10L19/0208

摘要： An audio signal coding device, an audio signal decoding device and a method to improve audio quality. The audio signal coding device and method include a quantizer that quantizes a given signal according to a number of assigned bits in order to generate a codeword. The coding device includes an extractor that extracts core bits from the generated codeword. The coding device also includes a determiner that determines an optimal value of the number of assigned bits based on an energy level corresponding to the extracted core bits. The audio signal decoding device and method include a dequantizer that dequantizes a given codeword according to the number of assigned bits to generate a decoded signal. The decoding device includes an extractor that extracts core bits from the given codeword. The decoding device also includes a determiner that determines an optimal value of the number of assigned bits used in the dequantizer.

摘要翻译： 音频信号编码装置，音频信号解码装置以及提高音频质量的方法。 音频信号编码装置和方法包括：量化器，用于根据分配的比特数来量化给定信号，以产生码字。 编码装置包括从所生成的码字中提取核心比特的提取器。 编码装置还包括确定器，其基于与所提取的核心比特相对应的能级确定分配比特数的最佳值。 该音频信号解码装置和方法包括一个去量化器，该去量化器根据分配的比特数量对给定的码字进行去量化，以产生解码的信号。 解码装置包括从给定码字中提取核心比特的提取器。 解码装置还包括确定器，其确定在去量化器中使用的分配位数的最佳值。

公开(公告)号：US09560456B2

公开(公告)日：2017-01-31

申请号：US13819349

申请日：2012-02-21

申请人： Yutaka Banba ,  Takeo Kanamori

发明人： Yutaka Banba ,  Takeo Kanamori

IPC分类号： H04R25/00 ,  G10L21/0208 ,  G10L21/0216

CPC分类号： H04R25/30 ,  G10L21/0208 ,  G10L2021/02165 ,  H04R25/407 ,  H04R2430/03

摘要： A hearing aid capable of detecting contact vibration noise from a collected sound signal. The hearing aid is provided with two microphones, a vibration component extracting section that extracts from collected sound signals respectively obtained by the two microphones an uncorrelated component between two collected sound signals as a vibration component for each frequency band. Additionally, a vibration noise identifying section determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting section, an acoustic signal processing section, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver converts the acoustic signal to sound.

摘要翻译： 一种能够从收集的声音信号中检测接触振动噪声的助听器。 助听器设置有两个麦克风，振动分量提取部分，从由两个麦克风分别获得的收集的声音信号中提取两个收集的声音信号之间的不相关分量作为每个频带的振动分量。 此外，振动噪声识别部分基于由振动分量提取部分提取的每个频带的振动分量来确定是否发生接触噪声，声学信号处理部分在通过辅助处理产生声学信号时确定 收集的声音信号，根据存在或不存在接触振动噪声的情况处理声信号，并且接收器将声信号转换成声音。

公开(公告)号：US09277316B2

公开(公告)日：2016-03-01

申请号：US13258171

申请日：2011-02-23

申请人： Yutaka Banba ,  Takeo Kanamori

发明人： Yutaka Banba ,  Takeo Kanamori

IPC分类号： H04R3/00 ,  H04R25/00

CPC分类号： H04R3/007 ,  H04R3/005 ,  H04R25/407 ,  H04R25/55 ,  H04R25/552 ,  H04R2225/41

摘要： A sound processing apparatus which can improve precision of analyzes on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors. The sound processing apparatus is provided with a level signal converter that converts the collected sound signal into a level signal, which indicates an absolute value of the collected sound signal from which phase information is removed. A level signal synthesizer generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors are synthesized, and a detector/identifier carries out analysis on the ambient sounds, based upon the synthesized level signal.

摘要翻译： 可以提高对环境声音的分析精度的声音处理装置，基于由两个声音收集器采集的声音信号对环境声音进行分析。 声音处理装置设置有电平信号转换器，该电平信号转换器将收集的声音信号转换为电平信号，该电平信号指示从其移除相位信息的所收集的声音信号的绝对值。 电平信号合成器产生合成电平信号，其中从两个收集器的收集的声音信号获取的电平信号被合成，并且检测器/标识符基于合成电平信号执行环境声音的分析。

公开(公告)号：US20120008797A1

公开(公告)日：2012-01-12

申请号：US13258171

申请日：2011-02-23

申请人： Yutaka Banba ,  Takeo Kanamori

发明人： Yutaka Banba ,  Takeo Kanamori

IPC分类号： H04R1/40

CPC分类号： H04R3/007 ,  H04R3/005 ,  H04R25/407 ,  H04R25/55 ,  H04R25/552 ,  H04R2225/41

摘要： A sound processing apparatus (100), which can improve precision of analyses on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors (first sound collector 110-1 and second sound collector 110-2), and the sound processing apparatus (100) is provided with a level signal conversion section (first level signal conversion section 130-1, second level signal conversion section 130-2) that converts the collected sound signal into a level signal, from which phase information is removed, a level signal synthesizing section (140) that generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors (first sound collector 110-1 and second sound collector 110-2) are synthesized, and a detecting and identifying section (160) that carries out analysis on the ambient sounds based upon the synthesized level signal.

摘要翻译： 能够提高环境声音分析精度的声音处理装置（100）基于由两个收集器（第一声音收集器110-1和第二声音收集器110-2）获取的收集的声音信号，对环境声音进行分析， ，并且声音处理装置（100）设置有将所收集的声音信号转换为电平信号的电平信号转换部（第一电平信号转换部130-1，第二电平信号转换部130-2），从该电平信号 信号被消除，电平信号合成部（140），其生成合成电平信号，其中从两个收集器（第一收集器110-1和第二声音收集器110-2）的收集的声音信号获取的电平信号是 以及基于合成电平信号对环境声音进行分析的检测和识别部分（160）。

公开(公告)号：US20090024395A1

公开(公告)日：2009-01-22

申请号：US10597215

申请日：2005-01-18

申请人： Yutaka Banba

发明人： Yutaka Banba

IPC分类号： G10L19/00

CPC分类号： G10L19/0204 ,  G10L19/083 ,  G10L2019/0003

摘要： It is an object of the present invention to provide an audio signal encoding method, an audio signal decoding method, a transmitter, a receiver, and a wireless microphone system which can compress an audio signal at a relatively high compression ratio at a relatively high quality with a relatively low delay. The compression encoder 4 of the transmitter includes an audio signal dividing filter bank 4a for dividing the audio signal into four sub-band signals, sampling each of the sub-band signals at a down-sampling rate depending on the number of the sub-band signals, and producing the sub-band signals sampled at the down-sampling rate, LD-CELP encoders 20a to 20d for encoding the sub-band signals on the basis of LD-CELP algorithm, and a multiplexer 4c for producing a multiplexed data stream with the encoded sub-band signals.

摘要翻译： 本发明的目的是提供音频信号编码方法，音频信号解码方法，发射机，接收机和无线麦克风系统，其可以以相对高的质量以相对高的压缩比压缩音频信号 具有相对较低的延迟。 发送器的压缩编码器4包括用于将音频信号分割为四个子带信号的音频信号分割滤波器组4a，其中，根据子带的数量，以下采样率对每个子带信号进行采样 信号，并产生以下采样率采样的子带信号，LD-CELP编码器20a至20d，用于根据LD-CELP算法对子带信号进行编码;以及多路复用器4c，用于产生多路复用数据流 与编码的子带信号。