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    • 22. 发明授权
    • Efficient computation of log-frequency-scale digital filter cascade
    • 对数倍频数字滤波器级联的高效计算
    • US07076315B1
    • 2006-07-11
    • US09534682
    • 2000-03-24
    • Lloyd Watts
    • Lloyd Watts
    • G06F17/00H04R25/00
    • H04R7/08
    • A system for processing audio signals comprises a sequence of digital filters each configured to process a selected frequency using a set of coefficients. A filter configured to process a certain frequency shares its coefficients with another filter that processes a frequency that is lower than the first frequency by at least one frequency interval, such as an octave. The first filter samples at a certain sampling rate, and the second filter's sampling rate is determined by multiplying the first sampling rate by the ratio of the second frequency to the first frequency. The filters are evenly grouped into frequency intervals, such as octaves. Filters in an octave are sampled at a sampling frequency that is at least twice as high as the highest frequency processed in that octave.
    • 用于处理音频信号的系统包括每个被配置为使用一组系数来处理选定频率的数字滤波器序列。 被配置为处理特定频率的滤波器与另一个滤波器共享其系数,该滤波器处理低于第一频率的频率至少一个频率间隔(例如八度)。 第一滤波器以一定的采样速率采样,第二滤波器的采样率通过将第一采样率乘以第二频率与第一频率的比来确定。 滤波器被均匀地分组成频率间隔,例如八度。 八度音阶中的滤波器采样频率至少是在该倍频程中处理的最高频率的两倍。
    • 23. 发明申请
    • Filter set for frequency analysis
    • 滤波器组用于频率分析
    • US20050216259A1
    • 2005-09-29
    • US10613224
    • 2003-07-03
    • Lloyd Watts
    • Lloyd Watts
    • H03H17/02H03H17/04G10L19/14
    • H03H17/02G10L25/18H03H17/04
    • A system and method are disclosed for analyzing an input signal into a plurality of frequency components. In one embodiment, the input signal is processed with a first set of low pass filters to derive a first set of frequency components wherein the first set of low pass filters are arranged serially in a chain having a first low pass filter and a last low pass filter, the output of each low pass filter being fed to the next low pass filter in the chain until the last low pass filter. The output of the last low pass filter is downsampled to produce a downsampled signal. The downsampled signal is processed with a second set of low pass filters to derive a second set of frequency components.
    • 公开了一种用于将输入信号分解为多个频率分量的系统和方法。 在一个实施例中,输入信号用第一组低通滤波器处理以导出第一组频率分量,其中第一组低通滤波器串联布置在具有第一低通滤波器和最后低通滤波器的链中 滤波器,每个低通滤波器的输出被馈送到链中的下一个低通滤波器,直到最后一个低通滤波器。 最后一个低通滤波器的输出被下采样以产生下采样信号。 下采样信号用第二组低通滤波器处理以导出第二组频率分量。
    • 24. 发明授权
    • Low-delay vector backward predictive coding of speech
    • 语音的低延迟向量反向预测编码
    • US4963034A
    • 1990-10-16
    • US360023
    • 1989-06-01
    • Vladimir M. CupermanRobert PettigrewLloyd Watts
    • Vladimir M. CupermanRobert PettigrewLloyd Watts
    • G10L19/00G10L19/12
    • G10L19/12G10L25/06
    • A method of encoding speech sounds to facilitate their transmission to and reconstruction at a remote receiver. A transmitter and a receiver have identical filters and identical codebooks containing prestored excitation vectors which model quantized speech sound vectors. The speech sound vectors are compared with filtered versions of the codebook vectors. The filtered vector closest to each speech sound vector is selected. During the comparison, filtration parameters derived by backward predictive analysis of a series of previously selected filtered codebook vectors are applied to the filter. The transmitter sends the receiver an index representative of the location of the selected vector within the codebook. The receiver uses the index to recover the selected vector from its codebook, and passes the recovered vector through its filter to yield an output signal which reproduces the original speech sound sample. By applying the same backward predictive analysis technique employed by the transmitter to the same series of previously selected filtered codebook vectors to which the transmitter applied the technique, the receiver derives the same combination of filtration parameters which the transmitter applied to its filter while selecting the codebook vector corresponding to the transmitted index.
    • 编码语音的方法,以便于它们在远程接收机处的传输和重构。 发射机和接收机具有相同的滤波器和相同的代码簿,其包含对量化语音声矢量进行建模的预存的激励矢量。 语音声音矢量与码本矢量的滤波版本进行比较。 选择最接近每个语音声矢量的滤波矢量。 在比较期间,通过一系列先前选择的滤波码本向量的反向预测分析得到的滤波参数被应用于滤波器。 发射机向接收机发送代表所选向量在码本内的位置的索引。 接收机使用索引从其码本中恢复所选择的向量,并将恢复的向量通过其滤波器,以产生再现原始语音声音样本的输出信号。 通过将发射机使用的相同的后向预测分析技术应用于发射机应用该技术的相同系列先前选择的滤波码本向量,接收机导出与发射机应用于其滤波器的滤波参数相同的组合,同时选择码本 向量对应于传输的索引。