CN106205595A - 一种用于个人音频设备的自适应噪声消除框架结构 - Google Patents

一种用于个人音频设备的自适应噪声消除框架结构 Download PDF

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CN106205595A
CN106205595A CN201610542543.5A CN201610542543A CN106205595A CN 106205595 A CN106205595 A CN 106205595A CN 201610542543 A CN201610542543 A CN 201610542543A CN 106205595 A CN106205595 A CN 106205595A
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J·D·亨德里克斯
G·D·卡马斯
N·卡瓦特拉
阿里·阿卜杜拉扎德米拉尼
杰弗里·奥尔德森
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Cirrus Logic Inc
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    • G10K11/17821Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase characterised by the analysis of input or output signals, e.g. frequency range, modes, transfer functions characterised by the analysis of the input signals only
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    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1787General system configurations
    • G10K11/17885General system configurations additionally using a desired external signal, e.g. pass-through audio such as music or speech
    • GPHYSICS
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    • G10K2210/1081Earphones, e.g. for telephones, ear protectors or headsets
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
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    • H04R1/00Details of transducers, loudspeakers or microphones
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Abstract

一种个人音频设备例如无线电话,包括自适应噪声消除(ANC)电路,其适用于自适应地从参考麦克风信号以及误差麦克风信号中产生抗噪信号,并且输入所述抗噪信号到扬声器换能器的输出以导致消除周围音频声音环境音频声音的消除,其中参考麦克风信号测量周围音频,并且误差麦克风测量换能器及其位置附近的环境音频声音的合成。处理电路使用所述参考麦克风以及误差麦克风以生成抗噪信号,其可以被由操作在ANC系数更新率的倍数上的自适应滤波器生成。下行音频可以通过插值与高数据率抗噪信号合成。在控制路径的高通滤波器减少ANC电路内的DC偏移,并且当下行音频没有被检测到时,ANC系数的适用可以被停止。

Description

一种用于个人音频设备的自适应噪声消除框架结构
本申请是分案申请,其母案申请为中国国家申请号201280027523.4,“一种用于个人音频设备的自适应噪声消除框架结构”,申请日2012年4月30日。
技术领域
本发明主要涉及包含自适应噪声消除(ANC)的个人音频设备例如无线电话,并且更具体地,涉及集成到个人音频设备中的ANC系统的体系结构特征。
背景技术
无线电话例如移动/便携电话、无绳电话,以及其他消费语音设备例如mp3播放器,被广泛使用。这些设备的清晰度相关的性能可以通过提供噪声消除而改进,改噪声消除使用麦克风测量周围环境事件并且然后使用信号处理以插入抗噪信号到设备的输出以消除环境声音事件。
由于个人音频设备例如无线电话周围的声音环境,取决于存在的噪声源以及设备本身的位置,可以剧烈变化,需要调适噪声消除以考虑这些环境变化。然而,自适应噪声消除电路可能变得复杂,消耗额外的能量,并且在特定环境下产生不必要的结果。
因此,需要提供一种包括无线电话的个人音频设备,其提供高效,能耗低,并且/或者复杂度低的噪声消除。
发明内容
通过在一种个人音频设备,操作方法以及集成电路中的实施,实现了提供了具有低能耗和/或低复杂度的噪声消除的个人音频设备。
所述个人音频设备包括壳体,其具有安装在壳体上的换能器,用于重现音频信号,该音频信号包括重放给听者的源音频以及用于补偿换能器声输出中的环境音频声音影响的抗噪信号;其可以包括集成电路以提供自适应噪声消除(ANC)功能。所述方法是操作上述个人音频设备以及集成电路的方法。参考麦克风被安装在壳体上,用于提供指示环境声音的参考麦克风信号。误差麦克风被包含用于控制抗噪信号的调适以消除环境音频声音并且用于修正从处理电路的输出到换能器的环境的电声路径。该个人音频设备进一步包括在壳体内的ANC处理电路,其用于一个或多个自适应滤波器适应地从参考麦克风信号以及参考麦克风中自适应地产生抗噪信号,从而该抗噪信号导致环境音频声音的大量消除。
ANC电路使用自适应滤波器,其生成能够工作在ANC系数更新率的倍数上的抗噪声信号。Σ-Δ调制器可以被包括在更高的采样率的信号路径(s)中以减小自适应滤波器(s)和其他处理模块的宽度。可以在控制路径中包含高通滤波器,以减少在ANC电路中的DC偏移,当下行音频不存在时,可以停止ANC的调适。当下行音频存在时,其可以合成高数据率的插值成的抗噪信号并且ANC的调适被恢复。
如附图中所描述的,从下述的对于本发明的优选实施例的具体描述,本发明前述以及其他的目标、特征以及优点将是十分清楚的。
附图说明
图1是依据本发明的实施例的无线电话10的示意图。
图2是依据本发明的实施例的无线电话10内的电路的框图。
图3是依据本发明的实施例的框图,描述了图2的编解码器集成电路20的ANC电路30内的信号处理电路以及功能模块。
图4是依据本发明的实施例的框图,描述了集成电路内信号处理电路以及功能模块。
图5是依据本发明的另一个实施例的框图,描述了集成电路内信号处理电路以及功能模块。
具体实施方式
本发明包括能够用于例如无线电话的个人音频设备中的噪声消除技术以及集成电路。该个人音频设备包括自适应噪声消除(ANC)电路,其测量周围声音环境并且产生一个信号,该信号被输入到扬声器(或者其它换能器)的输出以消除周围环境事件。提供了一个参考麦克风以测量周围声音环境并且包括一个误差麦克风用于控制抗噪信号的调适以消除环境音频声音并且用于修正从处理电路到换能器的电声路径。产生抗噪信号的自适应滤波器的系数控制可能在远低于自适应滤波器的样本频率的基带频率下工作,降低功率消耗和ANC处理电路的复杂性。高通滤波器可以被包括在提供输入给系数控制的反馈路径中,以在ANC控制回路降低直流(DC)偏移,并且当下行音频缺失时,ANC的自适应可能会停止,从而自适应滤波器的调适在可能导致不稳定的条件下不继续进行。当检测到下行音频,其可以通过在基带上提供以及与更高数据率音频插值而合成,自适应滤波器系数的调适被恢复。
现在参照图1,根据本发明示出的一个实施例无线中电话10靠近人耳5。所示出的无线电话10是依据本发明实施例能够使用的设备的一个示例,但是需要知道的是,不是无线电话10中所有的元件或者构造,或者后续说明中的电路,都是实施权利要求书所述的发明所需要的。无线电话10包括换能器例如扬声器SPKR,其用于重现无线电话所接收的远处语音以及其它本地的声音事件例如铃声、存储声音程序材料、近端语音输入(例如手机用户的语音)以提供平衡的会话感,以及其它需要声音无线电话10重现的声音,例如来自无线电话10接收的网页或者其它网络交互源的声音,以及语音提示例如低电量以及其它系统提示。近端麦克风NS被提供以捕捉近端语音,其被从无线电话10传输到其他会话参与者。
无线电话10包括自适应噪声消除(ANC)电路以及部件,其用于输入抗噪信号到扬声器SPKR以提高扬声器SPKR重现的远处语音以及其它语音的清晰度。参考麦克风R被提供用于测量周围声音环境,并且被放置在远离用户嘴巴的典型位置,从而近端语音在参考麦克风R产生的信号中被最小化。当无线电话10靠近于耳朵5时,提供第3麦克风,误差麦克风E,其通过提供将周围声音环境与临近耳朵5的麦克风SPKR合成的测量,用于进一步改进ANC操作。在无线电话10内的示范电路14包括声音编解码器集成电路20,其接收来自参考麦克风R,近语音麦克风NS以及误差麦克风E的信号并且与其它集成电路(例如包含无线电话收发器的RF合成电路12)交互。在本发明的其它实施例中,这里公开的电路和技术可以并入到单一的包含控制电路以及其它的功能的集成电路以用于实施个人语音设备的整体功能,例如MP3单芯片播放器(player-on-a-chip)的集成电路。
通常,本发明的ANC技术测量影响参考麦克风R的环境声音事件(不同于扬声器SPKR的输出和/或近端语音),并且也通过测量影响误差麦克风E的同样的环境声音事件,所述无线电话10的ANC处理电路调适由参考麦克风R输出产生的抗噪信号以形成一个特性,该特性最小化了呈现在误差麦克风E的周围环境信号的幅度。由于声音路径P(z)从参考麦克风R延伸到误差麦克风E,ANC电路实质上评估了与电声路径S(z)的消除效应合并的声音路径P(Z),电声路径S(z)表示编解码器IC20的音频输出电路的响应以及包括在特定的声音环境中扬声器SPKR与误差麦克风之间的耦合的扬声器SPKR的声/电转换性能,其在无线电话机10没有牢固地压到耳朵5时,被耳朵的周围和结构以及可能靠近无线电话10的其他物理对象以及人的头部结构影响。虽然所示的无线电话机10包括具有第三近语音麦克风NS的双麦克风ANC系统,据本发明的其它实施例,本发明的某些方面可以在不包括独立的误差和参考麦克风的系统中实施,或者本发明其它实施例中无线电话机使用近语音麦克风NS执行参考麦克风R的功能。同样,在只用于音频重放的个人音频设备中,通常不包括近语音麦克风,并且在电路中的近语音信号路径在下面的详述中可以忽略,而没有改变本发明的范围,而不是仅限于提供用于麦克风输入检测方案的选择。
现在参看图2,在无线电话10内的电路如框图所示。编解码器集成电路20包括模数字转换器(ADC)21A,用于接收参考麦克风信号和生成参考麦克风信号的数字表示ref,模数字转换器21B用于接收误差麦克风信号并且生成误差麦克风信号的数字表示err,以及模数字转换器21C用于接收近语音麦克风信号并且生成误差麦克风信号的数字表示ns。编解码器集成电路20生成用于从放大器A1驱动扬声器SPKR的输出,该放大器A1放大数模转换器(DAC)23的输出,该数模转换器(DAC)23接收合成器26的输出。合成器26合成音频信号,从内部音频源24、ANC电路30产生的抗噪信号(其通常具有与参考麦克风消耗ref中的噪声同样的极性,并且因此被合成器扣除)、近端语音信号ns的部分(以便无线电话10的用户听到适当对应于下行语音ds的自己的声音,该下行语音ds被从射频(RF)集成电路22接收并且同样被合成器26合成)。近语音信号ns也被提供给RF集成电路22,并作为上行语音通过天线ANT被发送到服务提供商。
现在参看图3,根据本发明的一个实施例示出ANC电路30的详情。自适应滤波器32接收参考麦克风信号ref并且在理想环境下,将其转换方程W(z)调适为P(z)/S(z)以产生抗噪信号,该抗噪信号被提供给输出合成器,该输出合成器将抗噪信号与换能器重现的音频合成,例如图2中示例的合成器26。自适应滤波器32的系数由W系数控制模块31控制,该W系数控制模块使用两个信号的关联来确定自适应滤波器32的响应,其通常在最小均方根意义上最小化出现在麦克风信号err中的参考麦克风信号ref的信号分量之间的误差。W系数控制模块31所比较的信号是由滤波器34B提供的响应路径S(z)的估计的副本所整形的参考麦克风信号ref,以及另外一个包含误差麦克风信号err的信号。通过使用响应路径S(z)的估计副本-响应SECOPY(z),转化参考麦克风信号ref,并且最小化合成信号和所述误差麦克风信号err之间的差分,自适应滤波器32调适为期望的P(z)/S(z)响应。下面详述的具有响应Cx(z)的滤波器37A,其处理滤波器34B的输出并且提供第一输出给W系数控制模块。W系数控制模块31的第二输入被另外一个具有的响应Ce(z)的滤波器37B处理。响应Ce(z)具有匹配于滤波器37A的响应Cx(z)的相位响应。滤波器37A和37B都具有高通响应,防止W(z)的系数受DC偏移以及极低频变化的影响。除了误差麦克风信号err以外,被W系数控制模块31与滤波器34B的输出相比的信号,包括已被滤波响应SE(z)处理的下行音频信号ds的反向量,其中响应SECOPY(z)是一个副本。通过输入下行音频信号ds的反向量,防止自适应滤波器32调适为误差麦克风信号err中存在的相对大量的下行音频,并且通过用响应路径S(z)的估计转换下行音频信号ds的反向副本,在比较前被从误差麦克风信号err中消除的下行音频应当匹配下行音频信号ds重现在误差麦克风信号err上的期望版本,因为S(z)的电学和声学路径是从下行音频信号ds开始到达误差麦克风E的路径。滤波器34B本身不是自适应滤波器,但其具有可调节的响应,该响应被调谐到匹配自适应滤波器34A的响应,从而使滤波器34B的响应跟踪自适应滤波器34A的调整。
为了实现上述,自适应滤波器34A具有由SE系数控制模块33控制的系数,其在消除上述被滤波的下行音频信号ds后,比较下行音频信号ds以及误差麦克风信号err,该被滤波的下行音频信号ds已经被自适应滤波器34A滤波以表示预期的被传送到误差麦克风E的下行音频,并且被合成器36A从自适应滤波器34A的输出中消除。SE系数控制模块33将实际下行语音信号ds与存在于误差麦克风信号err内的下行音频信号ds的分量关联。自适应滤波器34A从而调适为从下行音频信号ds成生一个信号,当该信号从误差麦克风信号err扣除时,该信号包含不是由下行音频信号产生的误差麦克风信号err的内容。下行音频检测模块39确定下行音频信号ds中何时包含信息,例如下行音频信号ds的水平大于阈值幅度。如果不存在下行音频信号DS,下行音频检测模块39使控制信号冻结有效,使得SE系数控制模块33和W系数控制模块31停止调适。
现在参照图4,显示了依据本发明的实施例的ANC技术的ANC系统的框图,其能够包含在图3描述的实施例中并且能够实施于图2的编解码器集成电路中。参考麦克风信号ref被Δ-ΣADC41A生成,该Δ-ΣADC41A操作64次过采样并且其输出由抽取器(decimator)42A的2因子抽取以产生32倍过采样信号。Δ-Σ整形器43A用于量化参考麦克风信号ref,其减少了后续处理级的宽度,例如滤波级44A和44B。由于滤波级44A和44B运行在过采样率上,Δ-Σ整形器43A能够将量化的噪声信号整形到量化噪声不产生干扰的频段,例如扬声器SPKR的频率响应波段之外,或者在其中电路系统的其他部分将不会通过量化噪声。滤波级44B具有固定的响应WFIXED(z),其通常被预先确定以在P(z)/S(z)的估计上提供起始点,用于特定用户的无线电话10的特定设计。P(z)/S(z)的估计响应的自适应部WADAPT(z)被自适应滤波级44A提供,该自适应滤波级44A被泄露(leaky)最小均方算法(LMS)系数控制器54A所控制。该泄露(leaky)最小均方算法(LMS)系数控制器54A是泄露的,由于当没有误差输入被提供以使泄露最小均方算法(LMS)系数控制器54A调适时,响应归一化到平整或者其它预定的响应。提供泄露控制器防止可能出现在一定的环境条件下的长期不稳定,并且通常使系统更能应对ANC响应的特定敏感度。
在图4所描绘的系统中,所述参考麦克风信号被具有响应SECOPY(z)的滤波器51滤波,响应SECOPY(z)是对响应路径S(z)的估计,其输出被抽取器(decimator)52A的32因子抽取以生成基带语音信号,该信号被通过无限脉冲响应(IIR)滤波器53A提供给泄露LMS54A。滤波器51本身不是自适应滤波器,但其具有可调节的响应,该响应被调谐到匹配滤波级55A和55B的合成响应,从而使滤波器51的响应跟踪响应SE(z)的调适。误差麦克风err由Δ-ΣADC41C产生,Δ-ΣADC41C操作于64倍过采样上并且其输出由抽取器42B的2因子抽取以产生32倍过采样信号。如图3的系统中,已被自适应滤波器使用响应SE(z)滤波的许多下行音频ds被合成器46C从误差麦克风信号err中消除,其输出由抽取器(decimator)53C通过32因子抽取以生成基带语音信号,该信号被通过IIR滤波器53B提供到泄露LMS54A。IIR滤波器53A和53B分别包括高通响应,防止DC偏移以及极低频变化影响自适应滤波器44A的系数的调适。
响应SE(z)被另一平行滤波级组55A和55B产生,其中一个滤波级55B具有固定响应SEFIXED(z),并且另外一个滤波级55A具有由泄露LMS系数控制器54B控制的自适应响应SEADAPT(z)。滤波器级55A和55B的输出由合成器46E合成。类似上述响应滤波器W(z)的实施,响应SEFIXED(z)通常是已知的预定响应以提供在各种工作条件下用于电/声路径S(Z)的适当起点。滤波器51是自适应滤波器55A/55B的副本,但其本身并不是自适应滤波器,即,滤波51不单独调适响应其本身的输出,并且滤波器51可以使用单级或双级实现。在图4的系统中提供了单独的控制值以控制滤波器51的响应,其中作为单级自适应滤波器级显示。然而,滤波器51可选地使用两个平行极实现,并且用于控制自适应滤波级55A的相同控制值也可以用于控制滤波器51方案中的可调滤波部。泄露LMS控制模块54B的输入同样是在基带,通过抽取合成器46H产生的下行语音信号ds和内部语音ia的合成而提供,该抽取由抽取器52B通过32因子抽取,并且另外一个输入由抽取合成器46C的输出提供,该输出已经消除由合成器46E合成的自适应滤波级55A和滤波级55B的输出而产生的信号。合成器46C的输出表示具有由于消除了下行音频信号ds的分量的误差麦克风信号err,其在被抽取器52C抽取后被提供到LMS控制模块54B。LMS控制模块54B的另一个输入是抽取器52B产生的基带信号。抽取器52B的输出上的下行音频信号ds(和内部音频信号ia)的大小被下行音频检测模块39检测,其在检测到下行音频信号ds和内部音频信号ia不存在时,冻结LMS控制模块54A54B的调适。
基带和过采样信号的上述设置,提供了简化控制并且减少自适应控制模块的消耗功率,如泄露LMS控制器54A和54B,同时提供在过采样率上实施自适应滤波器级44A-44B,55A-55B以及过滤器51产生的分接头灵活性。图4的系统的其余部分包括合成器46H,其将下行音频ds与内部音频ia合成,其输出被提供给合成器46D的输入,该合成器46D增加由Σ-ΔADC41B产生并且由侧音衰减器(attenuator)56滤波的近端麦克风信号ns以提供平衡的会话感。46D合成器的输出由Σ-Δ整形器43B整形,该Σ-Δ整形器43B提供输入到滤波级55A和55B,使用与上述Σ-Δ整形器43A同样的方式,使得滤波级55A和55B的宽度通过量化合成器46D的输出而缩小。Σ-Δ整形器43B的量化噪声被抽取器52C的内部低通响应移除。
依据本发明的实施例,合成器46D的输出同样与已被控制链处理过的自适应滤波器级44A-44B的输出合成,该控制链包括用于每个滤波级的相应硬静音模块(hard muteblock)45A、45B,合成硬静音模块45A、45B输出的合成器46A,软静音(soft mute)47以及还有软限制器(soft limiter)48以产生被合成器46B消除的抗噪信号,该合成器46B具有合成器46D的输出源音频。合成器46B的输出由插值器49的2因子插值,并且然后被操作在64倍过采样率的Σ-ΔDAC50重现。DAC50的输出被提供给放大器A1,其产生传送到扬声器SPKR的信号。
现在参看图5,示出根据另一个实施例的ANC技术的ANC系统的框图,可以被包括在本发明的图3所示的实施例中,如可以被实施在图2中编解码器集成电路20。图5中的ANC系统类似于图4,因此在下面将仅描述它们之间的区别。其不是提供高通滤波响应在泄漏LMS54A的输入上,而是通过在参考和误差麦克风信号路径提供各自的高通滤波器60A和60B,将DC分量直接从参考麦克风信号ref以及误差麦克风信号err中移除。然后额外的高通滤波器60C被包含滤波器51后的SE拷贝信号路径。图5中所示的体系结构的优点在于,高通滤波器60A从抗噪信号路径中移除分量和低频分量,否则其将在提供给扬声器SPKR的抗噪信号中通过滤波器级44A,44B,浪费能量,产生热量并且消耗动态范围。然而,由于参考麦克风信号ref需要包含一些能够被ANC系统消除的在低频段,即SPKR扬声器具有显着的响应的频率范围,的低频信息,滤波器60A被设计以通过这样的频率,而当用于泄漏LMS54A的优化调适时,更高的高通截止频率,例如,200赫兹,被使用。过滤器60B和60C的相位响应被匹配以保持用于泄漏LMS54A的稳定操作条件。
图4和图5系统中的每个或者部分单元,以及在图2和图3的示例电路中,可以直接使用逻辑电路,或由处理器例如数字信号处理(DSP)核心执行的程序指令实现,该程序指令执行例如自适应滤波和LMS系数计算的操作。虽然DAC和ADC极通常由专用混合信号电路实现,本发明的ANC系统的体系结构其本身将可以使用混合方式提供,其中逻辑电路可以例如在该设计的高度过采样部分使用,同时程序代码或微码驱动(microcode-driven)的处理单元被选择用于更复杂但较低速率的操作:如计算自适应滤波器的分接头和/或响应检测到的事件,例如本文所述的那些事件。
虽然参考其优选实施例的描述具体示出了本发明,但是应当明白本领域的技术人员在其中可以作出上述的和其它的形式和细节的变化而不脱离本发明的精神和范围。

Claims (6)

1.一种个人音频设备,包括:
个人音频设备壳体;
安装在壳体上的换能器,用于重现音频信号,包括重放给听者的源音频以及用于补偿换能器声输出中的环境音频声音影响的抗噪信号;
安装在壳体上的参考麦克风,用于提供指示环境音频声音的参考麦克风信号;
第一模数转换器,用于将参考麦克风信号以第一采样率转换到第一参考麦克风信号数字表示;
第一Σ-Δ量化器,其以所述第一采样率量化第一数字表示来以所述第一采样率生成降低了分辨率的第二参考麦克风信号数字表示;以及,
实施自适应滤波器的处理电路,该自适应滤波器具有从降低了分辨率的第二参考麦克风信号数字表示生成抗噪信号以减小听者听到的环境音频声音的存在的响应;其中,处理电路实施系数控制模块,其通过调适自适应滤波器的响应,将自适应滤波器的响应整形以一致于参考麦克风信号。
2.权利要求1所述的个人音频设备,其中:所述音频源是数字源音频表示,并且其中所述个人音频设备还包括:
第二Σ-Δ量化器,其量化数字源音频表示以生成降低分辨率的数字源音频表示;以及
安装在所述壳体上靠近所述换能器的位置的误差麦克风,用于提供指示所述换能器的声音输出以及所述换能器上环境音频声音的误差麦克风信号,其中处理电路实现了次级路径自适应滤波器以及合成器,该次级路径自适应滤波器具有将降低分辨率的数字源音频表示滤波的次级路径响应,该合成器从所述误差麦克风信号中消除滤波的所述源音频表示以提供误差信号到系数控制模块,该误差信号表示合成抗噪以及传输到听者的环境音频声音。
3.一种消除个人音频设备换能器附近的环境音频声音的方法,该方法包括:
使用参考麦克风测量环境音频声音的第一测量;
使用模数转换器将所述参考麦克风信号转换来以第一采样率产生第一参考麦克风数字表示;
使用Σ-Δ整形器将所述第一参考麦克风信号数字表示以所述第一采样率量化来以所述第一采样率生成降低分辨率的第二参考麦克风数字表示;以及
通过调适将降低分辨率的所述第二参考麦克风数字表示滤波的自适应滤波器的响应,自适应地从降低分辨率的所述第二参考麦克风信号数字表示生成抗噪信号以对抗在所述换能器的声输出的环境音频声音效应。
4.根据权利要求3所述的方法,还包括:
量化数字源音频表示以生成降低分辨率的数字源音频表示;以及
使用误差麦克风测量指示换能器的声输出以及换能器上环境音频声音的第二测量,其中所述自适应地生成包括:使用次级路径自适应滤波器将降低分辨率的数字源音频表示滤波,该次级路径自适应滤波器具有将所述降低分辨率的数字源音频表示整形的次级路径响应,以及从误差麦克风信号中消除次级路径自适应滤波器的输出结果以提供误差信号,该误差信号表示合成抗噪以及传输到听者的环境音频声音。
5.一种集成电路,用于执行个人音频设备的至少一部分,其包括:
输出,用于提供信号到换能器,包括重放给听者的源音频以及用于对抗换能器声输出中的环境音频声音影响的抗噪信号;
参考麦克风输入,用于接收指示环境音频声音的参考麦克风信号;
第一模数转换器,用于将参考麦克风信号以第一采样率转换到第一参考麦克风的数字表示;
第一Σ-Δ量化器,其以所述第一采样率量化第一数字表示来以所述第一采样率生成降低分辨率的第二参考麦克风信号数字表示;以及,
实现自适应滤波器的处理电路,该自适应滤波器具有从降低分辨率的第二参考麦克风信号数字表示生成抗噪信号以减小听者听到的环境音频声音的存在的响应;其中,所述处理电路实现系数控制模块,其通过调适自适应滤波器的响应,将自适应滤波器的响应整形以一致于参考麦克风信号。
6.权利要求5所述的集成电路,其中:音频源是数字源音频表示,并且其中所述集成电路还包括:
第二Σ-Δ量化器,其量化数字源音频表示以生成降低分辨率的数字源音频表示;以及
误差麦克风输入,用于接收指示所述换能器的声输出以及所述换能器上环境音频声音的误差麦克风信号,其中所述处理电路包括次级路径自适应滤波器以及合成器,该次级路径自适应滤波器具有将降低分辨率的数字源音频表示滤波以产生滤波后的源音频的次级路径响应,该合成器从误差麦克风信号中消除滤波的源音频表示以提供误差信号到系数控制模块,该误差信号表示合成抗噪以及传输到听者的环境音频声音。
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