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    • 4. 发明申请
    • METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNAL
    • 处理音频信号的方法和装置
    • US20160277865A1
    • 2016-09-22
    • US15031275
    • 2014-10-22
    • INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITYWILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE
    • Taegyu LEEHyunoh OHYoungcheol PARKDaehee YOUNJeongil SEOYongju LEESeungkwon BEACKKyeongok KANGDaeyoung JANG
    • H04S3/00G10L19/008
    • G10L19/008G10H2250/111G10H2250/145H04R3/00H04R5/033H04S3/00H04S3/002H04S3/004H04S3/008H04S2400/01H04S2420/01H04S2420/03
    • The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity.To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving truncated subband filter coefficients for filtering each subband signal of the input audio signal, the truncated subband filter coefficients being at least a portion of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal, the lengths of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, and the truncated subband filter coefficients being constituted by at least one FFT filter coefficient in which fast Fourier transform (FFT) by a predetermined block size in the corresponding subband has been performed; performing the fast Fourier transform of the subband signal based on a predetermined subframe size in the corresponding subband; generating a filtered subframe by multiplying the fast Fourier transformed subframe and the FFT filter coefficients; inverse fast Fourier transforming the filtered subframe; and generating a filtered subband signal by overlap-adding at least one subframe which is inverse fast Fourier transformed and an apparatus for processing an audio signal using the same.
    • 为此,提供了一种处理音频信号的方法,包括:接收输入音频信号; 接收用于对输入音频信号的每个子带信号进行滤波的截断的子带滤波器系数,截断的子带滤波器系数是从用于输入音频信号的双耳滤波的双耳室脉冲响应(BRIR)滤波器系数获得的子带滤波器系数的至少一部分, 基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息获得的滤波器顺序信息来确定截断的子带滤波器系数的长度,并且截断的子带滤波器系数由至少一个FFT滤波器系数构成,其中快速傅里叶 已经执行了在相应子带中的预定块大小的变换(FFT); 基于相应子带中的预定子帧大小执行子带信号的快速傅里叶变换; 通过乘以快速傅里叶变换子帧和FFT滤波器系数来产生滤波后的子帧; 对经滤波的子帧进行快速傅立叶逆变换; 以及通过重叠添加至少一个逆快速傅里叶变换的子帧和使用该子帧处理音频信号的装置来生成滤波后的子带信号。
    • 9. 发明授权
    • Method and apparatus for impulse response measurement and simulation
    • 用于脉冲响应测量和模拟的方法和装置
    • US09202450B2
    • 2015-12-01
    • US13554142
    • 2012-07-20
    • Mikko Pekka Vainiala
    • Mikko Pekka Vainiala
    • H04R29/00G10H1/16G10H3/18
    • G10H1/16G10H3/187G10H2210/311G10H2250/111H04R29/001
    • A method of measuring an impulse response of an amplifier coupled in operation to a loudspeaker arrangement includes: (a) coupling directly to a connection between the amplifier and the loudspeaker arrangement for obtaining access to a drive signal (Samp) applied to the loudspeaker arrangement to generate an acoustic output (S2); (b) disposing a microphone arrangement for receiving the acoustic output (S2) of the loudspeaker arrangement; (c) using a test signal generator to apply a test signal (Ssw) to an input of the amplifier; and (d) receiving at a digital signal processing arrangement (DSP, 210) at least the drive signal (Samp) and the acoustic output (S2) corresponding to the test signal (Ssw) and performing on these signals a signal processing operation for determining an impulse response for at least one of: the amplifier, the loudspeaker arrangement.
    • 测量在操作中耦合到扬声器装置的放大器的脉冲响应的方法包括:(a)直接耦合到放大器和扬声器装置之间的连接,以获得对应用于扬声器装置的驱动信号(Samp)的访问 产生声输出(S2); (b)设置用于接收扬声器装置的声输出(S2)的麦克风装置; (c)使用测试信号发生器将测试信号(Ssw)施加到放大器的输入端; 以及(d)至少在数字信号处理装置(DSP,210)处接收对应于测试信号(Ssw)的驱动信号(Samp)和声输出(S2),并对这些信号执行用于确定的信号处理操作 对于放大器,扬声器布置中的至少一个的脉冲响应。
    • 10. 发明授权
    • Measurement and processing of stringed acoustic instrument signals
    • 弦乐器信号的测量和处理
    • US06448488B1
    • 2002-09-10
    • US09889444
    • 2001-07-12
    • Ira EkhausLawrence Fishman
    • Ira EkhausLawrence Fishman
    • G10H106
    • G10H3/188G10H1/125G10H3/146G10H3/185G10H2220/395G10H2220/471G10H2220/501G10H2250/111G10H2250/235Y10S84/09
    • A system and a method of measuring, decomposing, processing and uniquely recombining forces and vibrations acting on stringed musical instruments (SMI). The system utilizes a digital signal processor and reproduces the musical sound characteristics of an acoustic instrument into high fidelity electrical signals for amplification, processing and/or filtering and reproduction of musical sounds by uniquely exploiting, through measurements and subsequent signal processing, the vector nature of string excitation forces (SEF) and body vibrations of stringed musical instruments. A signal processing system of the current invention also utilizes a plurality of sensors, each responsive to at least one of force, displacement, velocity or acceleration indicative of the vibrational energy of the strings, to produce a sensor signal vector, which is then processed and transformed by a plurality of re-creation filters into a transformed signal vector, and then resynthesized into an output signal. The resynthesized output signal be a microphone output signal, may have acoustic characteristics of another SMI or possess acoustic characteristics of a “theoretical” SMI.
    • 测量,分解,处理和独特重组作用在弦乐器(SMI)上的力和振动的系统和方法。 该系统利用数字信号处理器并将声学乐器的音乐声音特征再现为高保真电信号,用于通过测量和随后的信号处理独特地利用测量和随后的信号处理来扩展,处理和/或滤波和再现音乐声音 琴弦激发力(SEF)和弦乐器的身体振动。 本发明的信号处理系统还利用多个传感器,每个传感器响应于指示弦的振动能量的力,位移,速度或加速度中的至少一个,以产生传感器信号向量,然后将其传送到处理器 由多个再创建滤波器变换为变换信号矢量,然后再合成为输出信号。 重新合成的输出信号是麦克风输出信号,可以具有另一SMI的声学特性或具有“理论”SMI的声学特性。