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    • 11. 发明授权
    • Optimal looping for wavetable synthesis
    • 波形合成的最佳循环
    • US6084170A
    • 2000-07-04
    • US393442
    • 1999-09-08
    • Jean Laroche
    • Jean Laroche
    • G10H7/02G10H7/00G11C7/00
    • G10H7/02
    • In accordance with the present invention, a method and apparatus are provided wherein loop discontinuities are eliminated. In the case of amplitude discontinuities, the harmonic amplitudes contained in the loop are progressively scaled over the duration of the loop, so that for each harmonic the loop end amplitude matches the loop beginning amplitude. In the case of phase discontinuities, the harmonic phases are progressively shifted over the duration of the loop, so that for each harmonic the loop end phase matches the loop beginning phase. Shifting the phase is accomplished by slightly altering the frequency of the harmonics to produce the desired amount of phase-shift at the loop end. In accordance with the present invention, the method also provides a technique to select loop begin and end points to minimize the amount of phase adjustment.
    • 根据本发明,提供了一种消除循环不连续性的方法和装置。 在幅度不连续的情况下,循环中包含的谐波振幅在循环的持续时间内逐渐缩放,因此对于每个谐波,环路端振幅与环路起始幅度相匹配。 在相位不连续的情况下,谐波相位在环路的持续时间内逐渐移动,因此对于每个谐波,环路端相位与环路开始相位匹配。 通过稍微改变谐波的频率以在环路端产生期望量的相移来实现转换相位。 根据本发明,该方法还提供一种选择循环开始和结束点以最小化相位调整量的技术。
    • 12. 发明授权
    • Time-domain time/pitch scaling of speech or audio signals with transient
handling
    • 具有瞬态处理的语音或音频信号的时域时间/音调缩放
    • US6049766A
    • 2000-04-11
    • US745929
    • 1996-11-07
    • Jean Laroche
    • Jean Laroche
    • G10L21/04G10L3/02G10L9/00
    • G10L21/01
    • Method and apparatus for time-scaling and/or pitch shifting by discarding and/or repeating segments of a signal. The signal is stored as a series of samples in a memory where it is readable by one or more read pointers. Periodicity of segments of the signal is determined by evaluating normalized cross-correlation over a range of possible periods. Transients are detected by monitoring changes in rms signal value. To achieve time compression or time stretching, a segment is skipped/discarded whenever a maximum time-discrepancy between the current output and an ideal output is reached or a high periodicity is detected, a jump of the optimal length would not make this time discrepancy too high, and no transient is present in the segment to be skipped/discarded.
    • 用于通过丢弃和/或重复信号段来对时间缩放和/或音调移位的方法和装置。 该信号作为一系列样本存储在存储器中,其可被一个或多个读指针读取。 信号的段的周期性通过在可能的周期的范围内评估归一化的互相关来确定。 通过监测有效值信号值的变化检测瞬态。 为了实现时间压缩或时间延伸,每当达到当前输出和理想输出之间的最大时间差异或检测到高周期性时,跳过/丢弃一个段,最佳长度的跳跃也不会使此时间差异 高,并且在要跳过/丢弃的段中不存在瞬态。
    • 14. 发明授权
    • Multi-microphone active noise cancellation system
    • 多麦克风主动噪声消除系统
    • US08447045B1
    • 2013-05-21
    • US12876861
    • 2010-09-07
    • Jean Laroche
    • Jean Laroche
    • A61F11/06G10K11/16
    • A61F11/14A61F2011/145G10K11/178G10K2210/1081G10K2210/30232G10K2210/3055G10K2210/3219
    • The present technology provides systems and methods for robust feedforward active noise cancellation which can overcome or substantially alleviate problems associated with the diverse and dynamic nature of the surrounding acoustic environment. A multi-faceted analysis decouples the background noise within the earpiece from the acoustic wave (e.g. the anti-noise and desired audio) generated by an audio transducer within the earpiece. A difference signal is formed utilizing monitoring signals captured by array of monitoring microphones within the earpiece. The difference signal is formed such that contributions due to the acoustic wave generated by the audio transducer are selectively attenuated. As a result, the difference signal indicates an acoustic energy level of the background noise within the earpiece.
    • 本技术提供用于稳健的前馈有源噪声消除的系统和方法,其可以克服或基本上减轻与周围声环境的多种和动态特性相关的问题。 多面分析将耳机内的背景噪声与由耳机内的音频换能器产生的声波(例如抗噪声和期望音频)分离。 使用由耳机内的监视麦克风阵列捕获的监视信号形成差分信号。 差分信号形成为使得由音频换能器产生的声波引起的贡献被选择性地衰减。 结果,差信号表示耳机内的背景噪声的声能级。
    • 16. 发明授权
    • Synthesis of time-domain signals using non-overlapping transforms
    • 使用非重叠变换合成时域信号
    • US06311158B1
    • 2001-10-30
    • US09268878
    • 1999-03-16
    • Jean Laroche
    • Jean Laroche
    • G10L1304
    • G10L13/02
    • Techniques for synthesizing a time-domain signal. The time-domain signal is partitioned into a number of time-domain frames and a waveform in generated for each time-domain frame. Each waveform includes one or more sinusoids. The waveform is generated by selecting a sinusoid for synthesis and computing a set of parameter values (e.g. the start and end amplitude, frequency, and phase values) for the selected sinusoid. A template is determined for the selected sinusoid based on the computed parameter values and a selected window function. The frequency-domain template is such that the amplitude of the selected sinusoid in the time domain matches, at a time-domain frame boundary, the amplitude of a corresponding sinusoid in an adjacent time-domain frame. The template is added to a frequency-domain frame. The process is repeated for each sinusoid in the waveform. After all sinusoids have been processed, the frequency-domain frame is transformed to a time-domain frame. The time-domain frame is re-normalized with a re-normalization function that is generated based on the selected window function. A predetermined number of samples from each end of the time-domain frame can be discarded. The waveform is defined by the non-discarded samples in the time-domain frame. The waveforms from the time-domain frames are concatenated to generate the time-domain signal.
    • 用于合成时域信号的技术。 时域信号被划分成多个时域帧和针对每个时域帧产生的波形。 每个波形包括一个或多个正弦波。 通过选择用于合成的正弦波并计算所选择的正弦波的一组参数值(例如,起始和结束振幅,频率和相位值)来产生波形。 基于所计算的参数值和所选择的窗口函数,为所选择的正弦波确定模板。 频域模板使得时域中所选择的正弦波的振幅在时域帧边界处匹配相邻时域帧中相应正弦波的振幅。 模板被添加到频域框架中。 波形中的每个正弦波重复该过程。 在所有正弦曲线被处理之后,频域帧被转换成时域帧。 使用基于所选窗口函数生成的重新归一化函数对时域帧进行重新归一化。 可以丢弃来自时域帧的每一端的预定数量的采样。 波形由时域帧中的未丢弃样本定义。 来自时域帧的波形被级联以产生时域信号。
    • 17. 发明授权
    • Noise suppression assisted automatic speech recognition
    • 噪声抑制辅助自动语音识别
    • US09558755B1
    • 2017-01-31
    • US12962519
    • 2010-12-07
    • Jean LarocheCarlo Murgia
    • Jean LarocheCarlo Murgia
    • G10L21/02G10L21/00
    • G10L21/02G10L15/20G10L21/00G10L21/0232G10L25/78G10L2021/02165
    • Noise suppression information is used to optimize or improve automatic speech recognition performed for a signal. Noise suppression can be performed on a noisy speech signal using a gain value. The gain to apply to the noisy speech signal is selected to optimize speech recognition analysis of the resulting signal. The gain may be selected based on one or more features for a current sub band and time frame, as well as one or more features for other sub bands and/or time frames. Noise suppression information can be provided to a speech recognition module to improve the robustness of the speech recognition analysis. Noise suppression information can also be used to encode and identify speech.
    • 噪声抑制信息用于优化或改善对信号执行的自动语音识别。 噪声抑制可以使用增益值对噪声语音信号进行。 选择应用于噪声语音信号的增益以优化所得信号的语音识别分析。 可以基于用于当前子带和时间帧的一个或多个特征以及用于其他子带和/或时间帧的一个或多个特征来选择增益。 噪声抑制信息可以提供给语音识别模块,以提高语音识别分析的鲁棒性。 噪声抑制信息也可用于编码和识别语音。