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    • 2. 发明授权
    • Sound field and sound image control apparatus and method
    • 声场和声像控制装置及方法
    • US5684881A
    • 1997-11-04
    • US247269
    • 1994-05-23
    • Mitsuhiko SerikawaRyou TagamiAkihisa KawamuraMasaharu MatsumotoMikio OdaHiroko Numazu
    • Mitsuhiko SerikawaRyou TagamiAkihisa KawamuraMasaharu MatsumotoMikio OdaHiroko Numazu
    • H04S1/00H04N5/00H04B1/00
    • H04S1/007H04S1/002
    • 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.
    • 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。
    • 4. 发明授权
    • Distortion removal apparatus, method for determining coefficient for the same, and processing speaker system, multi-processor, and amplifier including the same
    • 失真去除装置,用于确定系数的方法,以及处理扬声器系统,多处理器和包括其的放大器
    • US06408079B1
    • 2002-06-18
    • US08932859
    • 1997-09-23
    • Taro KatayamaMitsuhiko Serikawa
    • Taro KatayamaMitsuhiko Serikawa
    • H03G500
    • H04R3/04H03G5/005H04R3/002
    • A distortion removal apparatus connected between a signal source and a signal input section of a distortion-generating system, for processing a signal output from the signal source so as to compensate for a distortion component generated in the system is provided. The distortion removal apparatus includes a frame division section for dividing a signal output from the signal source into data streams having a length of N while causing the data streams to partially overlap with one another; a Fourier transform section for performing Fourier transform of the data streams obtained by the frame division section in a time domain into a signal in a frequency domain; a memory section for storing N samples of first coefficients and N×N samples of second coefficients in the frequency domain. The distortion removal apparatus further includes an operation section for removing a distortion component from the output signal from the Fourier transform section by performing an operation based on the first coefficients and the output signal from the Fourier transform section and an operation based on the second coefficients and the output signal from the Fourier transform section; an inverse Fourier transform section for performing inverse Fourier transform of the output signal from the operation section into a signal in the time domain; and a frame synthesis section for sequentially connecting parts of the output signal from the inverse Fourier transform section.
    • 提供了一种连接在失真产生系统的信号源和信号输入部分之间的失真消除装置,用于处理从信号源输出的信号以补偿系统中产生的失真分量。 失真消除装置包括一个帧分割部分,用于将从信号源输出的信号分割为长度为N的数据流,同时使数据流彼此部分重叠; 傅里叶变换部分,用于在时域中对由帧划分部分获得的数据流进行频域的信号的傅立叶变换; 用于存储频域中的第一系数的N个采样和第二系数的N×N个采样的存储部分。 失真去除装置还包括一个操作部分,用于通过执行基于第一系数的操作和来自傅立叶变换部分的输出信号和基于第二系数的操作来从傅里叶变换部分的输出信号中去除失真分量,以及 来自傅里叶变换部的输出信号; 逆傅里叶变换部,用于将来自操作部的输出信号的傅立叶逆变换为时域中的信号; 以及帧合成部分,用于顺序地连接来自逆傅里叶变换部分的输出信号的部分。
    • 10. 发明授权
    • Audio signal compression method, audio signal compression apparatus, speech signal compression method, speech signal compression apparatus, speech recognition method, and speech recognition apparatus
    • 音频信号压缩方法,音频信号压缩装置,语音信号压缩方法,语音信号压缩装置,语音识别方法和语音识别装置
    • US06477490B2
    • 2002-11-05
    • US09892745
    • 2001-06-28
    • Yoshihisa NakatohTakeshi NorimatsuMineo TsushimaTomokazu IshikawaMitsuhiko SerikawaTaro KatayamaJunichi NakahashiYoriko Yagi
    • Yoshihisa NakatohTakeshi NorimatsuMineo TsushimaTomokazu IshikawaMitsuhiko SerikawaTaro KatayamaJunichi NakahashiYoriko Yagi
    • G10L1906
    • H04B1/665G10L2019/0005
    • An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
    • 一种用于对输入音频信号进行压缩编码的音频信号压缩装置包括用于将输入音频信号变换为频域信号的时间 - 频率变换单元; 频谱包络计算单元,用于根据输入的音频信号,使用基于人的听觉特征的频率的加权函数来计算用于不同频率的不同分辨率的频谱包络; 归一化单元,用于使用频谱包络对频域信号进行归一化以获得残余信号; 功率归一化单元,用于通过所述功率归一化所述残余信号; 听觉加权计算单元,用于基于输入音频信号的频谱和人类听觉特征来计算频率上的加权系数; 以及具有串联连接的多级矢量量化器的多级量化装置,其中输入归一化残差信号,以及使用加权系数量化残差信号的矢量量化器中的至少一个。 因此,与高频带相比,可以以更高的频率分辨率来分析具有重要意义的低频带,从而实现利用人类听觉特性的有效信号压缩。