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    • 6. 发明申请
    • ROBUST SPEECH BOUNDARY DETECTION SYSTEM AND METHOD
    • 鲁棒的语音边界检测系统和方法
    • US20140249812A1
    • 2014-09-04
    • US14197149
    • 2014-03-04
    • Conexant Systems, Inc.
    • Sahar E. Bou-GhazaleTrausti ThormundssonWillie B. Wu
    • G10L25/84G10L25/87
    • G10L25/84
    • A system for audio processing comprising an initial background statistical model system configured to generate an initial background statistical model using a predetermined sample size of audio data. A parameter computation system configured to generate parametric data for the audio data including cepstral and energy parameters. A background statistics computation system configured to generate preliminary background statistics for determining whether speech has been detected. A first speech detection system configured to determine whether speech was present in the initial sample of audio data. An adaptive background statistical model system configured to provide an adaptive background statistical model for use in continuous processing of audio data for speech detection. A parameter computation system configured to calculate cepstral parameters, energy parameters and other suitable parameters for speech detection. A speech/non-speech classification system configured to classify individual frames as speech frames or non-speech frames, based on the computed parameters and the adaptive background statistical model data. A background statistics update system configured to update the background statistical model based on detected speech and non-speech frames. A second speech detection system configured to perform speech detection processing and to generate a suitable indicator for use in processing audio data that is determined to include speech signals.
    • 一种用于音频处理的系统,包括初始背景统计模型系统,其被配置为使用音频数据的预定样本大小来生成初始背景统计模型。 参数计算系统,被配置为产生包括倒谱和能量参数的音频数据的参数数据。 背景统计计算系统,被配置为生成用于确定是否已经检测到语音的初步背景统计。 第一语音检测系统,被配置为确定音频数据的初始采样中是否存在语音。 一种自适应背景统计模型系统,被配置为提供用于语音检测的音频数据的连续处理中的自适应背景统计模型。 一种参数计算系统,被配置为计算语音检测的倒谱参数,能量参数和其他合适的参数。 一种语音/非语音分类系统,被配置为基于所计算的参数和自适应背景统计模型数据将各个帧分类为语音帧或非语音帧。 背景统计更新系统,被配置为基于检测到的语音和非语音帧来更新背景统计模型。 第二语音检测系统,被配置为执行语音检测处理并且生成用于处理被确定为包括语音信号的音频数据的合适的指示符。
    • 7. 发明申请
    • ADAPTIVE BIASING TECHNIQUE FOR AUDIO CIRCUITRY
    • 适用于音频电路的自适应偏移技术
    • US20140247091A1
    • 2014-09-04
    • US14197138
    • 2014-03-04
    • Conexant Systems, Inc.
    • Brian W. FriendLorenzo CrespiKyehyung Lee
    • H03F3/45
    • H03F1/0211H03F3/187H03F3/45179H03F3/45192H03F3/45475H03F2203/45112H03F2203/45494H03F2203/45528H03G1/0088
    • A circuit comprising a peak detector configured to receive a positive voltage input, a negative voltage input and a reference current source input and to output a peak signal data value. A fast attack current source control coupled to the peak detector and configured to generate a current source control signal as a function of the peak signal data value. A slow decay control coupled to the fast attack current source control and configured to reduce the current source control signal based on a predetermined or user-selected decay rate. A variable current source coupled to the fast attack current source control and configured to generate a variable current as a function of the current source control signal. Amplifier circuitry coupled to the variable current source, the amplifier circuitry configured to receive the variable current.
    • 一种电路,包括被配置为接收正电压输入的峰值检测器,负电压输入端和参考电流源输入端,并输出峰值信号数据值。 快速攻击电流源控制,耦合到峰值检测器并且被配置为产生作为峰值信号数据值的函数的电流源控制信号。 耦合到快速攻击电流源控制并被配置为基于预定或用户选择的衰减速率来减小电流源控制信号的慢衰减控制。 耦合到快速攻击电流源控制并被配置为根据电流源控制信号产生可变电流的可变电流源。 放大器电路耦合到可变电流源,放大器电路被配置为接收可变电流。
    • 8. 发明授权
    • Low area all digital delay-locked loop insensitive to reference clock duty cycle and jitter
    • 低区域全数字延迟锁定环对参考时钟占空比和抖动不敏感
    • US08779816B2
    • 2014-07-15
    • US13919973
    • 2013-06-17
    • Conexant Systems, Inc.
    • Santosh PatelPradeep Anantula
    • H03L7/06H03L7/08
    • H03L7/08H03L7/0805H03L7/0814H03L7/0816H03L7/093
    • A circuit comprising 1) a master delay-locked loop comprising a phase detector for receiving a reference clock and generating an output, control logic for receiving the output from the phase detector and a delta delay input and generating a control output, a clock splitter for receiving the reference clock and generating differential clock output, a delay line for receiving the differential reference clock from the clock splitter and generating n phases of differential reference clock at output, a multiplexer for receiving the output from the delay line and the control logic output and generating a clock output, wherein the phase detector is for receiving the reference clock, and 2) a slave delay-locked loop for receiving the control logic output and a strobe input and generating a delay locked loop output.
    • 一种电路,包括1)主延迟锁定环路,包括用于接收参考时钟并产生输出的相位检测器,用于接收来自相位检测器的输出的控制逻辑和增量延迟输入并产生控制输出的时钟分配器 接收参考时钟并产生差分时钟输出,延迟线,用于从时钟分离器接收差分参考时钟并在输出时产生差分参考时钟的n相;多路复用器,用于接收来自延迟线的输出和控制逻辑输出; 产生时钟输出,其中相位检测器用于接收参考时钟,以及2)用于接收控制逻辑输出和选通输入并产生延迟锁定环输出的从延迟锁定环路。
    • 9. 发明授权
    • Systems and methods for transducer calibration and tuning
    • 用于传感器校准和调谐的系统和方法
    • US08682002B2
    • 2014-03-25
    • US12828565
    • 2010-07-01
    • James Walter WihardjaMouna Elkhatib
    • James Walter WihardjaMouna Elkhatib
    • A61F11/06H04R29/00H03G5/00G06F7/00
    • H03G5/04H03G5/005H04R3/007H04R5/033H04R2420/07H04R2420/09
    • Audio transducers (headphones, speakers, microphones) inherently do not accurately reproduce the signal presented to them at the input. This can be compensated for by taking into account the transducer characteristics and transforming the input signal using a digital signal processor (DSP) to counteract the inaccuracies. However, for the compensation to take place, the DSP needs to know the characteristics of the transducer. For systems with built-in transducers (like laptops with internal speakers) the characteristics of the internal speakers can be stored on the hard-drive of the laptop and the DSP can read this data and make the appropriate compensations. Because a transducer (headphone, speaker, microphone) has its own characteristics that need to be compensated for separately, a profile is supplied to the DSP either by a database lookup based on an identification made by the user or transducer itself or by profile information stored on the transducer. Once the characteristics of a transducer are known, many additional DSP algorithms can be applied in order to improve the audio performance and even safety of the system.
    • 音频传感器(耳机,扬声器,麦克风)固有地不准确地再现在输入端呈现给它们的信号。 这可以通过考虑换能器特性并使用数字信号处理器(DSP)来变换输入信号来补偿,以抵消不准确性。 然而,为了进行补偿,DSP需要知道传感器的特性。 对于具有内置传感器(如具有内置扬声器的笔记本电脑)的系统,内部扬声器的特性可以存储在笔记本电脑的硬盘驱动器上,DSP可以读取此数据并进行适当的补偿。 因为传感器(耳机,扬声器,麦克风)具有需要单独补偿的自己的特性,所以通过基于由用户或换能器本身做出的标识的数据库查找或通过存储的简档信息将简档提供给DSP 在传感器上。 一旦知道传感器的特性,就可以应用许多附加的DSP算法,以便改善系统的音频性能甚至安全性。