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    • 56. 发明申请
    • SYSTEM AND METHOD FOR IDENTIFYING SUBOPTIMAL MICROPHONE PERFORMANCE
    • 用于识别低级麦克风性能的系统和方法
    • US20160050488A1
    • 2016-02-18
    • US14778643
    • 2013-03-21
    • Timo MATHEJAMarkus BUCKNUANCE COMMUNICATIONS, INC.
    • Timo MathejaMarkus Buck
    • H04R3/00
    • H04R3/00G06F3/165H04M1/03H04M1/6008H04R3/005H04R2430/03H04R2430/23H04R2499/11
    • Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.
    • 本文公开的实施例可以包括确定与计算设备相关联的第一麦克风和第二麦克风的信号参数。 实施例可以包括基于第一麦克风的参数和第二麦克风的参数中的至少一个来生成参考参数。 实施例可以包括基于参考参数调整第一麦克风和第二麦克风中的至少一个的容差。 实施例可以包括在第一麦克风处接收第一语音信号,第一语音信号具有第一语音信号幅度,并且在第二麦克风处接收第二语音信号,第二语音信号具有第二语音信号幅度。 实施例可以包括将第一语音信号幅度和第二语音信号幅度中的至少一个与第三语音信号幅度进行比较,并且基于该比较来检测阻塞的麦克风。
    • 57. 发明授权
    • Low-delay filtering
    • 低延迟滤波
    • US09036752B2
    • 2015-05-19
    • US14119933
    • 2011-05-05
    • Markus BuckTobias Wolff
    • Markus BuckTobias Wolff
    • H04B1/00H04B1/10H03H17/02H03H21/00G10L21/0232
    • H04B1/10G10L21/0232H03H17/0213H03H21/0027
    • A method of frequency-domain filtering is provided that includes a plurality of filters, the plurality of filters including at least one constrained filter(s) W=I, I and at least one unconstrained filter(s) W=1,K− The method includes cascading the W k=i,K unconstrained filter(s). A single constraint window C is applied to the cascaded W=i,K unconstrained filter(s). The W=1,I constrained filter(s) are cascaded with the constrained cascaded Wk=1,K unconstrained filter(s) to form a resulting filter Wll=C(W 1{circle around (x)} . . . {circle around (x)} W){circle around (x)} W . . . W. The frequency domain representation of the single constraint window C may be based, at least in part, on a time domain representation of a single constraint window C that has been circularly shifted such that the frequency domain representation of the constraint window matches a property of the frequency domain representation of the cascaded W=1,K unconstrained filters.
    • 提供了包括多个滤波器的频域滤波的方法,所述多个滤波器包括至少一个约束滤波器W = I,I和至少一个无约束滤波器W = 1,K- 方法包括级联W k = i,K无约束滤波器。 单个约束窗口C被应用于级联的W = i,K无约束滤波器。 W = 1,约束滤波器与受限级联Wk = 1,K无约束滤波器级联,以形成滤波器W11 = C(W 1 {围绕(x)}圆圈...圆圈 around(x)} W){circle around(x)} W。 。 。 单个约束窗口C的频域表示可以至少部分地基于已被循环移位的单个约束窗口C的时域表示,使得约束窗口的频域表示与属性 的级联W = 1的频域表示,K个无约束滤波器。
    • 58. 发明授权
    • System for speech signal enhancement in a noisy environment through corrective adjustment of spectral noise power density estimations
    • 通过频谱噪声功率密度估计的校正调整,在噪声环境中进行语音信号增强的系统
    • US08364479B2
    • 2013-01-29
    • US12202147
    • 2008-08-29
    • Gerhard Uwe SchmidtTobias WolffMarkus Buck
    • Gerhard Uwe SchmidtTobias WolffMarkus Buck
    • G10L21/02
    • H04R3/00G10L21/0208G10L21/0216
    • A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.
    • 系统估计音频信号的频谱噪声功率密度包括频谱噪声功率密度估计单元,校正项处理器和组合处理器。 频谱噪声功率密度估计单元可以提供音频信号的频谱噪声功率密度的第一估计。 校正项处理器可以至少部分地基于实际频谱噪声功率密度的频谱噪声功率密度估计误差来提供时间相关校正项。 可以确定校正项,使得谱噪声功率密度估计误差降低。 组合处理器可以将第一估计与校正项组合以获得可用于后续信号处理以增强音频信号的期望信号分量的频谱噪声功率密度的第二估计。
    • 59. 发明授权
    • Determination of the coherence of audio signals
    • 确定音频信号的相干性
    • US08238575B2
    • 2012-08-07
    • US12636432
    • 2009-12-11
    • Markus BuckTimo Matheja
    • Markus BuckTimo Matheja
    • H04B15/00
    • G10L25/78G10L2021/02165
    • Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals.
    • 本发明的实施例公开了用于估计信号一致性的计算机实现的方法,系统和计算机程序产品。 首先,由第一麦克风检测由声源产生的声音以获得第一麦克风信号,并由第二麦克风检测第二麦克风信号。 第一麦克风信号被第一自适应有限脉冲响应滤波器滤波以获得第一滤波信号。 第二麦克风信号被第二自适应有限脉冲响应滤波器滤波,以获得第二滤波信号。 基于经滤波的信号确定第一滤波信号和第二滤波信号的相干性。 第一麦克风信号和第二麦克风信号被滤波,以便在声音传输到第一麦克风的声音传递功能与声音从声源传输到第二麦克风之间的差异被补偿在 第一和第二滤波信号。