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    • 1. 发明申请
    • ENHANCED BLIND SOURCE SEPARATION ALGORITHM FOR HIGHLY CORRELATED MIXTURES
    • 用于高相关混合物的增强型盲源分离算法
    • WO2009097413A1
    • 2009-08-06
    • PCT/US2009/032414
    • 2009-01-29
    • QUALCOMM INCORPORATEDWANG, SongRAMAKRISHNAN, DineshGUPTA, Samir KumarCHOY, Eddie L.T.
    • WANG, SongRAMAKRISHNAN, DineshGUPTA, Samir KumarCHOY, Eddie L.T.
    • H04R3/00H04R25/00G10L21/02
    • G10L21/028G10L2021/02166H04R3/005H04R25/40
    • An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages.
    • 提供增强的盲源分离技术来改善高度相关的信号混合物的分离。 波束形成算法用于预处理相关的第一和第二输入信号,以避免通常与盲源分离相关联的不确定性问题。 波束成形算法可以对第一信号和第二信号应用空间滤波器,以便在衰减来自其它方向的信号的同时放大来自第一方向的信号。 这种方向性可以用于在第一信号中放大期望的语音信号,并从第二信号中衰减所需的语音信号。 然后对波束形成器输出信号执行盲源分离,以分离所需的语音信号和环境噪声,并重构所需语音信号的估计。 为了增强波束形成器和/或盲源分离的操作,可以在一个或多个阶段执行校准。
    • 3. 发明申请
    • DIGITAL DOMAIN SAMPLING RATE CONVERTER
    • 数字域采样率转换器
    • WO2007095624A2
    • 2007-08-23
    • PCT/US2007/062250
    • 2007-02-15
    • QUALCOMM INCORPORATEDWANG, SongCHOY, Eddie L.T.KULKAMI, Prajakt, V.GUPTA, Samir, Kumar
    • WANG, SongCHOY, Eddie L.T.KULKAMI, Prajakt, V.GUPTA, Samir, Kumar
    • H03H17/06
    • H03H17/0685H03H17/0294
    • Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter. A sampling rate converter up-samples the digital signal at an input sampling frequency to the selected intermediate sampling frequency, filters the digital signal with the derived anti-aliasing filter, and down-samples the digital signal by the selected down-sampling factor to the desired output sampling frequency.
    • 通过根据选择的中间采样频率对数字信号进行上采样和下采样来描述数字域中的采样率转换的技术。 具有多种因素带宽的原型抗混叠滤波器存储在内存中。 该技术包括基于原型滤波器的因子来选择中间采样频率为数字信号的期望输出采样频率的整数倍,并且选择下采样因子为与所选择的中间采样相关联的相同整数 频率。 滤波器生成器基于原型滤波器为选定的下采样因子生成抗混叠滤波器。 采样率转换器将输入采样频率的数字信号上采样到选定的中间采样频率,用导出的抗混叠滤波器对数字信号进行滤波,并将数字信号通过选择的下采样因子下采样到 所需的输出采样频率。
    • 4. 发明申请
    • AUTOMATIC VOLUME AND DYNAMIC RANGE ADJUSTMENT FOR MOBILE AUDIO DEVICES
    • 自动音量和动态范围调整移动音频设备
    • WO2008134689A1
    • 2008-11-06
    • PCT/US2008/061947
    • 2008-04-30
    • QUALCOMM IncorporatedXIANG, PeiWANG, SongKULKARNI, Prajakt VGUPTA, Samir KumarCHOY, Eddie L.t.
    • XIANG, PeiWANG, SongKULKARNI, Prajakt VGUPTA, Samir KumarCHOY, Eddie L.t.
    • H03G3/32H04M1/60
    • H03G7/007H03G3/32H04M1/6016
    • A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) (125) and Automatic Volume Control (AVC) (126) to increase the volume of sound (127) emitted from a speaker of' the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) (133) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.
    • 移动音频设备(例如,蜂窝电话,个人数字音频播放器或MP3播放器)执行音频动态范围控制(ADRC)(125)和自动音量控制(AVC)(126)以增加声音音量(127 )从“移动音频设备”的扬声器发出,使得音频的微弱通道更可听见。 这种微弱通道的放大发生,而不会过度放大其他更大的通道,并且没有由于限幅导致的实质性变形。 例如,多麦克风有源噪声消除(MMANC)(133)功能用于从移动音频设备的麦克风拾取的音频信息中去除背景噪声。 然后可以从设备传送噪声消除的音频。 MMANC功能产生噪声参考信号作为中间信号。 中间信号被调节,然后用作AVC处理的参考。 在AVC过程中应用的增益是噪声参考信号的函数。