会员体验
专利管家(专利管理)
工作空间(专利管理)
风险监控(情报监控)
数据分析(专利分析)
侵权分析(诉讼无效)
联系我们
交流群
官方交流:
QQ群: 891211   
微信请扫码    >>>
现在联系顾问~
热词
    • 1. 发明申请
    • Multi-channel audio signal limiter with shared clip detection
    • 具有共享剪辑检测功能的多声道音频信号限制器
    • US20040081324A1
    • 2004-04-29
    • US10279355
    • 2002-10-24
    • Kai Kwong LauRobert Kelly CadenaJohn Elliott Whitecar
    • H04B015/00H03G007/00
    • H03G7/002
    • An audio system limits gain in individual audio channels having a shared clip detect signal from a multi-channel power amplifier such that only channels likely to be exceeding the distortion threshold are gain limited. In addition, any arbitrary channels may be grouped together for common gain limiting. The invention monitors the power level of each audio channel, compares the power level to a power threshold that indicates whether a high power condition with the potential to cause excess distortion exists or not, and makes a decision whether to activate each channel's gain limiter based on whether the corresponding clip detect signal is active and the high power condition exists simultaneously for that channel.
    • 音频系统限制具有来自多声道功率放大器的共享剪辑检测信号的各个音频通道中的增益,使得仅有可能超过失真阈值的通道被增益受限制。 此外,任何任意通道可以被组合在一起用于共同的增益限制。 本发明监视每个音频通道的功率电平,将功率电平与功率阈值进行比较,该功率阈值指示是否存在具有引起超量失真的可能性的高功率状态,并且基于以下方式决定是否激活每个通道的增益限制器 相应的片段检测信号是否有效,并且该信道同时存在高功率条件。
    • 4. 发明申请
    • Binaural compression system
    • 双耳压缩系统
    • US20040190734A1
    • 2004-09-30
    • US10353187
    • 2003-01-27
    • GN ReSound A/S
    • James M. Kates
    • H04B001/64H03G007/00
    • H04B1/64H03G7/06H04R25/356H04R25/552
    • A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs. The multi-channel signal processing system further includes peak detecting means detecting signal peaks of the first and second channel input signals for generating first and second channel control signals. Thereafter, gain adjusting means adjusts signal gains of the first and second channel control signals. The first and second channel compressing units then respectively compress the first and second channel input signals to produce the first and second channel compressed outputs in accordance with the adjusted first and second channel control signals, respectively.
    • 提供了一种适用于提供音调输入的双耳压缩的多声道信号处理系统。 这样的系统可以用于例如双耳助听器系统中以提供音调输入的动态范围双耳压缩。 多通道信号处理系统本质上是一种具有通过两个信号通道之间的控制链路连接的两个信号通道的系统,从而允许双耳助听器系统模拟诸如人类的交叉橄榄色丛(COCB)效应的行为 听觉系统包括左耳和右耳之间的神经连接。 多信道信号处理系统包括分别位于多信道信号处理系统的第一和第二信号信道中的第一和第二信道压缩单元。 第一和第二信道压缩单元分别接收第一和第二信道输入信号,以产生第一和第二信道压缩输出。 多通道信号处理系统还包括检测第一和第二信道输入信号的信号峰值的峰值检测装置,用于产生第一和第二信道控制信号。 此后,增益调整装置调整第一和第二信道控制信号的信号增益。 第一和第二信道压缩单元然后分别压缩第一和第二信道输入信号,以分别根据调整的第一和第二信道控制信号产生第一和第二信道压缩输出。
    • 5. 发明申请
    • Sharing wavelet domain components among encoded signals
    • 在编码信号之间共享小波域分量
    • US20040071300A1
    • 2004-04-15
    • US10269894
    • 2002-10-10
    • Texas Instruments Incorporated
    • Daniel L. ZelazoSteven D. Trautmann
    • H03G007/00
    • G06F17/148G10L19/0216
    • A system for sharing wavelet domain components among encoded signals receives a set of signals decomposed and encoded according to a wavelet transform. The decomposed and encoded signals each include a set of wavelet coefficients at each level of the decomposition of the encoded signal. Using a vector quantization technique, the system identifies one or more sets of wavelet coefficients that are sharable among two or more of the decomposed and encoded signals at a particular level of decomposition. The system then stores the sets of wavelet coefficients of the decomposed and encoded signals. Each identified sharable set of wavelet coefficients at a particular level of decomposition is stored only once and shared by two or more of the decomposed and encoded signals.
    • 用于在编码信号中共享小波域分量的系统接收根据小波变换分解和编码的一组信号。 分解和编码的信号各自包括在编码信号的分解的每个级别处的一组小波系数。 使用矢量量化技术,系统识别在特定分解级别的两个或多个分解和编码信号之间可共享的一组或多组小波系数。 然后,系统存储分解和编码信号的小波系数集合。 在特定分解级别的每个识别的可共享小波系数集合仅被存储一次并由两个或更多个分解和编码信号共享。
    • 6. 发明申请
    • Method for noise reduction of a FM signal
    • 降低FM信号噪声的方法
    • US20030039371A1
    • 2003-02-27
    • US10225849
    • 2002-08-22
    • Jens Wildhagen
    • H04H005/00H03G007/00
    • H04H20/48H04B1/1692H04H40/45
    • A compander for noise reduction of a FM signal is described, wherein a group delay (null) linked to the generation of the compressor gain (cc(t)) is equalised during generation of the multiplex signal (m(t)), and a group delay (null) linked to the generation of the expander gain (ce(t)) is equalised during generation of the sum signal (us(t)) and the expanded difference signal (ue(t)). Alternatively or additionally the compressor gain and/or the expander gain is controlled by an auxiliary signal on the basis of a combination of sum signal and difference signal of the FM signal. Such companders avoid overmodulation in the transmitter.
    • 描述了用于FM信号的降噪的压缩扩展器,其中在生成多路复用信号(m(t))期间与压缩机增益(cc(t))的生成相关联的组延迟(τ)被均衡,并且 在和信号(us(t))和扩展差分信号(ue(t))的产生期间,与扩展器增益(ce(t))的产生相关联的组延迟(τ)被均衡。 或者或另外,基于和信号和FM信号的差分信号的组合,通过辅助信号来控制压缩机增益和/或扩展器增益。 这种压缩器避免了发射机中的过调制。
    • 7. 发明申请
    • Efficient digital audio automatic gain control
    • 高效数字音频自动增益控制
    • US20020181724A1
    • 2002-12-05
    • US09828116
    • 2001-04-06
    • Zhongnong JiangJames R. Hochschild
    • H04B001/64H03G007/00H04R025/00
    • H03G7/007
    • The present invention is a digital dynamic compression or automatic gain control (AGC) (10) adapted for use in high quality audio and hearing aids applications. An efficient digital AGC design employs two compact ROM-based tables (ROM_CSD, ROM_SPL) in addition to two comparators (COMP_A, COMP_B) and several registers (REG_A, REG_B, ADDR_A, ADDR_B). While one ROM stores the values of discrete input signal levels, the other contains gain codes based on a canonical signed digit (CSD) coding approach that leads to a very simple gain multiplier (20). In many cases an extremely compact table for gain values can be achieved by reusing a single small-size ROM that behaves like one that is several time larger. Two design examples are shown to expound the insights of the new digital AGC design. For the less-than-half-dB-gain-step cases only two adders are required for the multiplier whereas just three adders are needed in the situations with less than quarter-dB gain steps.
    • 本发明是适用于高质量音频和助听器应用的数字动态压缩或自动增益控制(AGC)(10)。 除了两个比较器(COMP_A,COMP_B)和几个寄存器(RE​​G_A,REG_B,ADDR_A,ADDR_B)之外,高效的数字AGC设计采用两个紧凑的基于ROM的表(ROM_CSD,ROM_SPL)。 虽然一个ROM存储离散输入信号电平的值,另一个包含基于经典有符号数字(CSD)编码方法的增益代码,其导致非常简单的增益乘法器(20)。 在许多情况下,增益值非常紧凑的表格可以通过重新使用一个像数倍更大的那样的单个小型ROM来实现。 展示了两个设计实例来阐述新的数字AGC设计的见解。 对于小于1/2 dB的增益步长情况,乘法器只需要两个加法器,而在具有小于四分之一dB增益步长的情况下,只需要三个加法器。
    • 8. 发明申请
    • Dynamic range compression of an audio signal
    • 音频信号的动态范围压缩
    • US20020126860A1
    • 2002-09-12
    • US09758679
    • 2001-01-11
    • Clinton A. Staley
    • H03G007/00
    • H03G7/007
    • Computer processor method and apparatus for creating an audio multiplier control signal for controlling the dynamic range of a recorded audio work. The technique includes determining an envelope of the amplitude of amplitude versus time values the audio signal, and then determining, for values of the envelope, respective minimum and maximum multiplication factors (MinMF and MaxMF) that can be multiplied times the values such that the products are above a predetermined minimum amplitude and below a predetermined maximum amplitude of the dynamic range. Then a control signal function of amplitude versus time is created such that all values of the control signal function at particular times are between respective MinMF and MaxMF values for the times, and such that segments of the control signal function have reduced slopes. Also disclosed is a method of creating a reduced-slope series of line segments passing through a pair of Max and Min limiting functions specifying y values with respect to a variable x.
    • 用于创建用于控制记录的音频作品的动态范围的音频乘法器控制信号的计算机处理器方法和装置。 该技术包括确定音频信号的振幅对时间值的包络,然后对于包络值确定可以乘以乘以值的各自的最小和最大乘法因子(MinMF和MaxMF),使得乘积 高于预定的最小幅度并且低于动态范围的预定最大幅度。 然后,创建幅度对时间的控制信号功能,使得控制信号在特定时间功能的所有值在时间之间在相应的MinMF和MaxMF值之间,并且使得控制信号功能的段具有减小的斜率。 还公开了一种创建通过一组Max和Min限制函数的减少斜率系列的线段的方法,所述Max和Min限制函数相对于变量x指定y值。