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    • 2. 发明授权
    • Factorial packing method and apparatus for information coding
    • 信息编码的因子包装方法和装置
    • US06236960B1
    • 2001-05-22
    • US09370057
    • 1999-08-06
    • Weimin PengEdgardo Manuel Cruz ZenoJames Patrick Ashley
    • Weimin PengEdgardo Manuel Cruz ZenoJames Patrick Ashley
    • G10L2104
    • G10L19/10
    • An improved speech coder takes advantage of the fact that any given pulse combination can be uniquely described by the following four properties: number of degenerate pulses, signs of pulses, positions of pulses, and pulse magnitudes. In accordance with the invention, a four stage iterative classification of the pulse combinations, where each stage groups the pulse combinations by one of these four properties, is performed. The process starts with the number of pulses, then determines the total number of possible sign combinations, pulse position combinations, and pulse magnitude combinations. This flexibility allows for the sign combinations to be grouped in the last stage. Since the number of sign combinations is always a power of two, leaving the sign combinations for last along with appropriately ordering the elements in the previous three stages allows the signs to be coded by independent bits, in turn allowing for error protection of those bits.
    • 改进的语音编码器利用以下事实:任何给定的脉冲组合可以通过以下四个特性唯一描述:简并脉冲数,脉冲符号,脉冲位置和脉冲幅度。 根据本发明,执行脉冲组合的四阶段迭代分类,其中每个阶段通过这四个属性之一来组合脉冲组合。 该过程以脉冲数开始,然后确定可能的符号组合的总数,脉冲位置组合和脉冲幅度组合。 这种灵活性允许在最后阶段将符号组合分组。 由于符号组合的数量总是2的幂,所以最后留下符号组合以及前三个阶段中的元素的适当排序允许符号被独立的位编码,从而允许这些位的错误保护。
    • 3. 发明授权
    • Method and apparatus for suppressing acoustic background noise in a communication system by equaliztion of pre-and post-comb-filtered subband spectral energies
    • 通过前后梳状滤波的子带频谱能量的均衡来抑制通信系统中的声学背景噪声的方法和装置
    • US06366880B1
    • 2002-04-02
    • US09451074
    • 1999-11-30
    • James Patrick Ashley
    • James Patrick Ashley
    • G10L2102
    • G10L21/0208
    • A noise suppression system implemented in communication system provides an improved level of quality during severe signal-to-noise ratio (SNR) conditions. The noise suppression system, inter alia, incorporates a frequency domain comb-filtering (289) technique which supplements a traditional spectral noise suppression method. The invention includes a real cepstrum generator (285) for an input signal (285) G(k) to produce a likely voiced speech pitch lag component and converting a result to frequency domain to obtain a comb-filter function (290) C(k), applying input signal (291) G(k) to comb-filter function (290) C(k), and equalizing the energies of the corresponding pre and post filtered subbands, to produce a signal (293) G″(k) to be used for noise suppression. This prevents high frequency components from being unnecessarily attenuated, thereby reducing muffling effects of prior art comb-filters.
    • 在通信系统中实现的噪声抑制系统在严重的信噪比(SNR)条件下提供了改善的质量水平。 噪声抑制系统尤其包含频域梳状滤波(289)技术,其补充传统的频谱噪声抑制方法。 本发明包括用于输出信号(285)G(k)的真实倒谱发生器(285),以产生可能的有声语音音调滞后分量并将结果转换到频域以获得梳状滤波器函数(290)C(k ),将输入信号(291)G(k)应用于梳状滤波器函数(290)C(k),并且对相应的前和后滤波后的子带的能量进行均衡,以产生信号(293)G“ )用于噪声抑制。 这防止高频分量被不必要地衰减,从而减少现有技术梳状滤波器的消音效果。
    • 4. 发明授权
    • Method and apparatus for coding and decoding speech
    • 用于语音编码和解码的方法和装置
    • US06415252B1
    • 2002-07-02
    • US09086396
    • 1998-05-28
    • Weimin PengJames Patrick Ashley
    • Weimin PengJames Patrick Ashley
    • G10L1106
    • G10L19/04G10L25/93G10L2019/0004
    • Bits are allocated to short-term repetition information for unvoiced input signals. Stated differently, more bits are allocated for pitch information during unvoiced input speech than in the prior art. The improved method and apparatus in an encoder (300) and decoder (700) result in improved consistency of amplitude pulses compared to prior art methods which indicates improved stability due to increased search resolution. Also, the improved method and apparatus result in higher energy compared to prior art methods which indicates that the synthesized waveform matches the target waveform more closely, resulting in a higher fixed codebook (FCB) gain.
    • 位被分配给无声输入信号的短期重复信息。 换句话说,在清音输入语音中比现有技术更多的比特被分配用于音调信息。 与现有技术方法相比,编码器(300)和解码器(700)中改进的方法和装置导致振幅脉冲的一致性提高,这表明由于增加的搜索分辨率而提高了稳定性。 此外,与现有技术方法相比,改进的方法和装置导致更高的能量,这表明合成波形更加紧密地匹配目标波形,导致更高的固定码本(FCB)增益。
    • 8. 发明授权
    • Method and apparatus for estimating the fundamental frequency of a signal
    • 用于估计信号的基频的方法和装置
    • US06188979B1
    • 2001-02-13
    • US09086509
    • 1998-05-28
    • James Patrick Ashley
    • James Patrick Ashley
    • G10L1914
    • G10L25/90G10L19/09
    • A method and apparatus for improved pitch period (&tgr;) estimation in a compression system is disclosed. The system uses original estimates of integer lag (&tgr;0) and open-loop prediction gain (&bgr;ol) as input to an adaptive filter parameter initialization block (304) which supplies inputs to a plurality of adaptive filter elements (306-308). Adaptive filter elements (306-308) provide information regarding the harmonics of the residual signal (&egr;(n)) to an adaptive filter parameter analysis block (310). Adaptive filter parameter analysis block (310) estimates the fundamental frequency of the residual signal based on the analysis of the harmonics and outputs a pitch period (&tgr;) for eventual use in a delay contour computation.
    • 公开了一种用于改进压缩系统中的音调周期(&tgr)估计的方法和装置。 系统使用整数滞后(&tgr; 0)的原始估计和开环预测增益(βOL)作为向多个自适应滤波器元件(306-308)提供输入的自适应滤波器参数初始化块(304)的输入。 自适应滤波器元件(306-308)将关于残余信号(epsi(n))的谐波的信息提供给自适应滤波器参数分析块(310)。 自适应滤波器参数分析块(310)基于谐波分析估计残余信号的基频,并输出用于延迟轮廓计算中最终使用的音调周期(&tgr)。
    • 9. 发明授权
    • Method and apparatus for coding an information signal
    • 用于对信息信号进行编码的方法和装置
    • US06141638A
    • 2000-10-31
    • US86149
    • 1998-05-28
    • Weimin PengJames Patrick Ashley
    • Weimin PengJames Patrick Ashley
    • G10L19/00G10L19/10G10L19/14G10L9/00
    • G10L19/18G10L19/10
    • A speech coder (400) for coding an information signal varies the codebook configuration based on parameters inherent in the information signal. The speech coder (400) requires no additional overhead for sending of mode parameters while allowing subframe resolution. The configurations vary not only for voicing level, but also for pitch period since different physiological traits yield different codebook configurations. A dispersion matrix (406) within the speech coder (400) facilitates a codebook search which is performed on vectors whose length can be less than a subframe length. Additionally, use of the dispersion matrix (406) allows the addition of random events for very slightly voiced speech which incurs little computational overhead but produces a rich excitation.
    • 用于编码信息信号的语音编码器(400)根据信息信号中固有的参数来改变码本配置。 语音编码器(400)不需要用于发送模式参数的额外开销,同时允许子帧分辨率。 这些配置不仅对于发声水平而言也是变化的,而且对于音调周期也是不同的,因为不同的生理特征产生不同的码本配置。 语音编码器(400)内的分散矩阵(406)便于对长度可以小于子帧长度的矢量执行的码本搜索。 此外,使用色散矩阵(406)允许为非常轻微的语音语音添加随机事件,这引起很少的计算开销,但产生了丰富的激励。