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1.发明授权Method of encoding and decoding speech using modified logarithmic transformation with offset of line spectral frequency 失效使用线谱频率偏移的改进对数变换对语音进行编码和解码的方法
公开(公告)号：US6131083A
公开(公告)日：2000-10-10
申请号：US219773
申请日：1998-12-23
申请人： Kimio Miseki , Katsumi Tsuchiya
发明人： Kimio Miseki , Katsumi Tsuchiya
CPC分类号： G10L19/07
摘要： On the basis of an autocorrelation coefficient calculated by an autocorrelation coefficient computation section from an input speech signal, an LSF computation section computes LSF parameters F(k) (k=1, 2, . . . , N). A modified logarithmic transformation section performs on the LSF parameters a logarithmic transformation with offset defined by f(k)=logC (1+A.times.F(k)) to obtain modified logarithmic LSF parameters f(k). The resulting modified logarithmic LSF parameters are quantized by a quantization section to provide quantized LSF parameters fq(k). Codes representing the quantized LSF parameters fq(k) are outputted. An inverse transformation defined by Fq(k)=(C.sup.fq(k) -1)/A is performed on the LSF parameters fq(k) to output LSF parameters Fq(k) on the general frequency scale.
摘要翻译： 基于自相关系数计算部分从输入语音信号计算的自相关系数，LSF计算部分计算LSF参数F（k）（k = 1,2，...，N）。 改进的对数变换部分对LSF参数执行由f（k）= logC（1 + AxF（k））定义的偏移的对数变换，以获得修正的对数LSF参数f（k）。 所得到的修正对数LSF参数由量化部分量化，以提供量化的LSF参数fq（k）。 输出表示量化的LSF参数fq（k）的代码。 对LSF参数fq（k）执行由Fq（k）=（Cfq（k）-1）/ A定义的逆变换，以在一般频率标度上输出LSF参数Fq（k）。
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2.发明授权Speech coding method and apparatus for coding an input speech signal based on whether the input speech signal is wideband or narrowband 有权基于输入语音信号是宽带还是窄带来编码输入语音信号的语音编码方法和装置公开(公告)号：US08260621B2
公开(公告)日：2012-09-04
申请号：US12751421
申请日：2010-03-31
申请人： Kimio Miseki
发明人： Kimio Miseki
CPC分类号： G10L19/18
摘要： A wideband speech coding apparatus comprises a wideband speech coding unit configured to code an input speech signal at a bit rate which is set in advance in accordance with a wideband speech signal. The apparatus further comprises an identification unit configured to identify whether the input speech signal is a wideband speech signal or a narrowband speech signal. The apparatus further comprises a control unit configured to cause the wideband speech coding unit to code the input speech signal, when the identification unit identifies that the input speech signal is the wideband speech signal, and to cause the wideband speech coding unit to raise the bit rate to code the input speech signal, when the identification unit identifies that the input speech signal is the narrowband speech signal. The apparatus further comprises an output unit to output the input speech signal coded by the wideband speech coding unit.
摘要翻译： 宽带语音编码装置包括：宽带语音编码单元，被配置为以根据宽带语音信号预先设置的比特率对输入语音信号进行编码。 该装置还包括识别单元，其被配置为识别输入的语音信号是宽带语音信号还是窄带语音信号。 该装置还包括：控制单元，配置为当所述识别单元识别出所述输入语音信号是所述宽带语音信号时，使所述宽带语音编码单元对所述输入语音信号进行编码，并且使所述宽带语音编码单元提升所述比特 速率来编码输入语音信号，当识别单元识别出输入语音信号是窄带语音信号时。 该装置还包括输出单元，用于输出由宽带语音编码单元编码的输入语音信号。
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公开(公告)号：US08244547B2
公开(公告)日：2012-08-14
申请号：US12549659
申请日：2009-08-28
申请人： Takashi Sudo , Kimio Miseki
发明人： Takashi Sudo , Kimio Miseki
IPC分类号： G10L19/00
CPC分类号： G10L21/038
摘要： A signal bandwidth extension apparatus includes a determination unit which determines whether or not a peak component of the input signal is lacked in the band to be extended, and a control unit which controls to extend the bandwidth when the determination unit determines that the peak component of the input signal is lacked in the band to be extended, and not to extend the bandwidth when the determination unit determines that the peak component is not lacked.
摘要翻译： 信号带宽扩展装置包括确定单元，其确定在要扩展的频带中输入信号的峰值分量是否不足;以及控制单元，其在所述确定单元确定所述峰值分量 在要延长的频带中缺少输入信号，并且当确定单元确定不存在峰值分量时不延长带宽。
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公开(公告)号：US08238568B2
公开(公告)日：2012-08-07
申请号：US12888321
申请日：2010-09-22
CPC分类号： H04R3/04
摘要： According to one embodiment, an acoustic signal compensator includes an acoustic signal receiving module, a compensator, and an output module. The acoustic signal receiving module receives an acoustic signal. The compensator performs compensation on the acoustic signal, as compensation of acoustic characteristics of an ear including an ear canal having a first-order resonance characteristic and a second-order resonance characteristic, to suppress a first-order frequency of ear resonance and a second-order frequency lower than double of the first-order frequency. The output module outputs the acoustic signal compensated by the compensator.
摘要翻译： 根据一个实施例，声信号补偿器包括声信号接收模块，补偿器和输出模块。 声信号接收模块接收声信号。 补偿器对声信号执行补偿，作为包括具有一阶谐振特性和二次谐振特性的耳道的耳朵的声学特性的补偿，以抑制耳谐振的一阶频率， 订单频率低于一阶频率的两倍。 输出模块输出由补偿器补偿的声信号。
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公开(公告)号：US20100250245A1
公开(公告)日：2010-09-30
申请号：US12751421
申请日：2010-03-31
申请人： Kimio Miseki
发明人： Kimio Miseki
IPC分类号： G10L21/00
CPC分类号： G10L19/18
摘要： A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.
摘要翻译： 一种宽带语音编码方法，包括识别输入语音信号是窄带信号还是宽带信号，以及通过基于识别结果控制宽带语音编码处理的预定参数来对输入语音信号进行编码。
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公开(公告)号：US07788105B2
公开(公告)日：2010-08-31
申请号：US11240495
申请日：2005-10-03
申请人： Kimio Miseki
发明人： Kimio Miseki
CPC分类号： G10L19/18
摘要： A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.
摘要翻译： 一种宽带语音编码方法，包括识别输入语音信号是窄带信号还是宽带信号，以及通过基于识别结果控制宽带语音编码处理的预定参数来对输入语音信号进行编码。
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公开(公告)号：US20080279394A1
公开(公告)日：2008-11-13
申请号：US12012976
申请日：2008-02-06
申请人： Takehiko Isaka , Kimio Miseki
发明人： Takehiko Isaka , Kimio Miseki
IPC分类号： H04B15/00
CPC分类号： H04S1/00
摘要： A common component extraction unit extracts a component Mi(n) included commonly in the plural input channels. A non-common component extraction unit extracts a component Si(n) not common to the plural input channels. A common component noise suppression unit obtains signal Mo(n) by executing a noise suppression process for the component Mi(n). A non-common component processing unit obtains signal So(n) by executing attenuation or emphasis for the component Si(n). A plural-channel generation unit removes noise of the common component from signals In(n, k) of the plural input channels, by using the signals Mo(n) and So(n) such that the non-common component is attenuated or emphasized.
摘要翻译： 公共分量提取单元提取通常包含在多个输入通道中的分量Mi（n）。 非公共分量提取单元提取多个输入通道不共同的分量Si（n）。 公共分量噪声抑制单元通过对分量Mi（n）执行噪声抑制处理来获得信号Mo（n）。 非公共分量处理单元通过对分量Si（n）执行衰减或强调来获得信号So（n）。 多信道生成单元通过使用信号Mo（n）和So（n）从多个输入信道的信号In（n，k）中除去公共分量的噪声，使得非共同分量被衰减或强调 。
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公开(公告)号：US20080161952A1
公开(公告)日：2008-07-03
申请号：US11810995
申请日：2007-06-07
申请人： Masataka Osada , Hirokazu Takeuchi , Kimio Miseki
发明人： Masataka Osada , Hirokazu Takeuchi , Kimio Miseki
CPC分类号： G10L19/008
摘要： According to an aspect of the invention, there is provided an audio data processing apparatus including: a decoding unit configured to decode audio encoding data, while the decoding unit switches an M/S stereo application mode and an M/S stereo non-application mode, and thereby outputting frequency domain audio data; an inverse quantizing unit configured to inversely quantize and output the frequency domain audio data; an M/S stereo judgment unit configured to decide whether or not the M/S stereo application mode is applied to the scale factor band, and extract and output a frequency domain audio data of the S channel at a part of scale factor band to which the M/S stereo application mode is applied, and generate and output a frequency domain audio data of the S channel at a part of scale factor band to which the M/S stereo application mode is not applied.
摘要翻译： 根据本发明的一个方面，提供了一种音频数据处理装置，包括：解码单元，被配置为解码音频编码数据，而解码单元切换M / S立体声应用模式和M / S立体声非应用模式 从而输出频域音频数据; 反量化单元，被配置为对频域音频数据进行逆量化和输出; M / S立体声判断单元，其被配置为判定是否将M / S立体声应用模式应用于比例因子频带，并且在比例因子频带的一部分提取并输出S频道的频域音频数据， 应用M / S立体声应用模式，并且在不应用M / S立体声应用模式的比例因子频带的一部分处生成并输出S频道的频域音频数据。
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9.发明授权Speech encoding and decoding with pitch filter range unrestricted by codebook range and preselecting, then increasing, search candidates from linear overlap codebooks 失效语音编码和解码与音调滤波器范围不受码本范围和预选择限制，然后增加从线性重叠码本搜索候选
公开(公告)号：US5819213A
公开(公告)日：1998-10-06
申请号：US791741
申请日：1997-01-30
申请人： Masahiro Oshikiri , Tadashi Amada , Masami Akamine , Kimio Miseki
发明人： Masahiro Oshikiri , Tadashi Amada , Masami Akamine , Kimio Miseki
CPC分类号： G10L19/08
摘要： A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.
摘要翻译： 一种语音编码方法和装置，包括使用在预定搜索范围内表达语音参数的码本来分析与音码周期相对应的音频周期的声音加权滤波器中的输入语音信号，并且从码本中搜索 基于分析结果，将输入语音信号的失真最小化的语音参数的组合以及编码该组合。 该装置使用音调和噪声码本的自适应码本。 码本搜索通过从一个原始码矢量提取预定长度的矢量而形成的组，同时依次移位位置使得矢量彼此重叠。 搜索组进一步限制，并在最终搜索之前进行另一预选。 搜索基于逆卷积正交变换载体。
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公开(公告)号：US08577051B2
公开(公告)日：2013-11-05
申请号：US13093601
申请日：2011-04-25
CPC分类号： H04R3/04
摘要： According to one embodiment, a sound signal compensation apparatus includes an input module, a compensation module, and an output module. The input module receives identification information identifying a first frequency with regard to a resonance of an ear closed by an earphone or headphone. The compensation module performs first compensation emphasizing a second frequency on a sound signal, the second frequency being determined based on the identification information or the first frequency. The output module outputs the compensated sound signal. The compensation module is configured to perform the first compensation emphasizing the second frequency, at which emphasis is greater than or equal to 2 dB and less than or equal to 12 dB.
摘要翻译： 根据一个实施例，声音信号补偿装置包括输入模块，补偿模块和输出模块。 输入模块接收关于由耳机或耳机关闭的耳朵的共振识别第一频率的识别信息。 补偿模块执行强调声音信号的第二频率的第一补偿，第二频率是基于识别信息或第一频率确定的。 输出模块输出补偿后的声音信号。 补偿模块被配置为执行强调第二频率的第一补偿，其中强调大于或等于2dB且小于或等于12dB。
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