公开(公告)号：US06717991B1

公开(公告)日：2004-04-06

申请号：US09493265

申请日：2000-01-28

申请人： Harald Gustafsson ,  Ulf Lindgren ,  Ingvar Claesson ,  Sven Nordholm

发明人： Harald Gustafsson ,  Ulf Lindgren ,  Ingvar Claesson ,  Sven Nordholm

IPC分类号： H04B1500

CPC分类号： H04R3/005

摘要： Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal. Finally, a third spectral subtraction function is used to enhance the near-mouth signal by suppressing the background noise using the enhanced background noise estimate. A controller dynamically determines any or all of a first, second, and third subtraction factor for each of the first, second, and third spectral subtraction stages, respectively.

摘要翻译： 通过使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法，在双麦克风降噪系统中提供语音增强。 根据示例性实施例，当远端麦克风与近嘴麦克风结合使用时，只要可以从单个输入样本块连续地估计噪声谱，就可以处理非平稳背景噪声。 远端麦克风除了吸收背景噪音外，还会吸收扬声器的声音，尽管距离近嘴麦克风较低。 为了增强噪声估计，使用频谱减法级来抑制远口麦克风信号中的语音。 为了能够增强噪声估计，粗略的语音估计是由近端信号的另一个谱减法阶段形成的。 最后，使用第三频谱减法函数，通过使用增强的背景噪声估计来抑制背景噪声来增强近嘴信号。 控制器分别动态地确定第一，第二和第三频谱减法级中的每一个的第一，第二和第三减法因子中的任何一个或全部。

公开(公告)号：US06175602B1

公开(公告)日：2001-01-16

申请号：US09084387

申请日：1998-05-27

申请人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

发明人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC分类号： H04B1500

CPC分类号： G10L21/0208

摘要： Methods and apparatus for providing speech enhancement in noise reduction systems include spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, low order spectrum estimates are developed which have less frequency resolution and reduced variance as compared to spectrum estimates in conventional spectral subtraction systems. The low order spectra are used to form a gain function having a desired low variance which in turn reduces musical tones in the spectral subtraction output signal. Advantageously, the gain function can be further smoothed across blocks using input spectrum dependent exponential averaging. Additionally, the low order of the gain function permits a phase to be added during interpolation so that the spectral subtraction gain filter is causal and prevents discontinuities between blocks.

摘要翻译： 用于在降噪系统中提供语音增强的方法和装置包括使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法。 根据示例性实施例，与常规频谱减法系统中的频谱估计相比，开发出具有较低频率分辨率和减小的方差的低阶频谱估计。 低阶光谱用于形成具有期望的低方差的增益函数，这进而减小谱减法输出信号中的乐音。 有利地，可以使用输入频谱相关的指数平均来使得增益函数在块之间进一步平滑。 另外，增益函数的低阶允许在插值期间添加相位，使得谱减法增益滤波器是因果关系并且防止块之间的不连续性。

公开(公告)号：US06507623B1

公开(公告)日：2003-01-14

申请号：US09289555

申请日：1999-04-12

申请人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

发明人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC分类号： H04B1500

CPC分类号： G10L21/0208

摘要： For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. Moreover, since the spectral subtraction gain function is computed in a block-wise fashion in the frequency domain, high quality performance in terms of reduced tonal artifacts and low signal distortion is retained.

摘要翻译： 为了噪声抑制的目的，在时域中以采样方式执行频谱减法滤波，使用在频域中以块方式计算的频谱减法增益函数的时域表示。 通过以样本为基础连续执行时域滤波，所公开的方法和装置避免了与基于频域的谱减法系统相关的块处理延迟。 因此，所公开的方法和装置特别适用于需要非常短的处理延迟的应用。 此外，由于在频域中以块方式计算频谱减法增益函数，所以保留了降低的色调伪影和低信号失真方面的高质量性能。

公开(公告)号：US06549586B2

公开(公告)日：2003-04-15

申请号：US09289065

申请日：1999-04-12

申请人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

发明人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC分类号： H04B1500

CPC分类号： H04R3/005

摘要： Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal. Finally, a third spectral subtraction function is used to enhance the near-mouth signal by suppressing the background noise using the enhanced background noise estimate.

摘要翻译： 通过使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法，在双麦克风降噪系统中提供语音增强。 根据示例性实施例，当远端麦克风与近嘴麦克风结合使用时，只要可以从单个输入样本块连续地估计噪声谱，就可以处理非平稳背景噪声。 远端麦克风除了吸收背景噪音外，还会吸收扬声器的声音，尽管距离近嘴麦克风较低。 为了增强噪声估计，使用频谱减法级来抑制远口麦克风信号中的语音。 为了能够增强噪声估计，粗略的语音估计是由近端信号的另一个谱减法阶段形成的。 最后，使用第三频谱减法函数，通过使用增强的背景噪声估计来抑制背景噪声来增强近嘴信号。

公开(公告)号：US06487257B1

公开(公告)日：2002-11-26

申请号：US09289554

申请日：1999-04-12

申请人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

发明人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC分类号： H04B1500

CPC分类号： G10L21/0208

摘要： For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. In applications where only stationary, low-energy background noise is present, computational complexity is reduced by generating a number of separate spectral subtraction gain functions during an initialization period, each gain function being suitable for one of several predefined classes of input signal (e.g., for one of several predetermined signal energy ranges), and thereafter fixing the several gain functions until the input signal characteristics change.

摘要翻译： 为了噪声抑制的目的，在时域中以采样方式执行频谱减法滤波，使用在频域中以块方式计算的频谱减法增益函数的时域表示。 通过以样本为基础连续执行时域滤波，所公开的方法和装置避免了与基于频域的谱减法系统相关的块处理延迟。 因此，所公开的方法和装置特别适用于需要非常短的处理延迟的应用。 在仅存在静止的低能量背景噪声的应用中，通过在初始化周期期间产生多个单独的频谱减法增益函数来减少计算复杂度，每个增益函数适用于若干预定义类别的输入信号之一（例如， 对于几个预定信号能量范围中的一个），然后固定多个增益函数直到输入信号特性改变。

公开(公告)号：US06459914B1

公开(公告)日：2002-10-01

申请号：US09084503

申请日：1998-05-27

申请人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

发明人： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC分类号： B13

CPC分类号： G10L21/0232 ,  G10L21/0264

摘要： Methods and apparatus for providing speech enhancement in noise reduction systems include spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, successive blocks of a spectral subtraction gain function are averaged based on a discrepancy between an estimate of a spectral density of a noisy speech signal and an averaged estimate of a spectral density of a noise component of the noisy speech signal. The successive gain function blocks are averaged, for example, using controlled exponential averaging. Control is provided, for example, by making a memory of the exponential averaging inversely proportional to the discrepancy. Alternatively, the averaging memory can be made to increase in direct proportion with decreases in the discrepancy, while exponentially decaying with increases in the discrepancy to prevent audible voice shadows.

摘要翻译： 用于在降噪系统中提供语音增强的方法和装置包括使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法。 根据示例性实施例，基于噪声语音信号的频谱密度的估计与噪声语音信号的噪声分量的频谱密度的平均估计之间的差异来平均频谱减法增益函数的连续块。 连续增益功能块被平均化，例如使用受控指数平均。 提供控制，例如，通过使指数平均值与差异成反比的记忆。 或者，可以使平均存储器随着差异的减小而成正比地增加，同时随着差异的增加呈指数衰减，以防止可听见的声音阴影。

公开(公告)号：US06549627B1

公开(公告)日：2003-04-15

申请号：US09016264

申请日：1998-01-30

申请人： Jim Rasmusson ,  Ingvar Claesson ,  Mattias Dahl ,  Sven Nordholm

发明人： Jim Rasmusson ,  Ingvar Claesson ,  Mattias Dahl ,  Sven Nordholm

IPC分类号： H04R300

CPC分类号： H04M9/082 ,  G01S7/537 ,  H04R1/406 ,  H04R3/005 ,  H04R2201/401

摘要： A beamformer is calibrated for use as an acoustic echo canceler in a hands-free communications environment having a loudspeaker and a plurality of microphones. To perform the calibration, a number of adaptive filters are provided in correspondence with each of the microphones, and each of the adaptive filters is trained to model echo properties of the environment as experienced by the corresponding one of the microphones. A target source is activated, thereby generating an acoustic signal that is received by the microphones. The trained adaptive filters are then used to generate jammer signals by, for example, having each one filter a pseudo noise signal. Respective ones of the jammer signals are then combined with corresponding signals supplied by the microphones, thereby generating combination signals. The combination signals are then used to adapt the beamformer to cancel the jammer signals. In another aspect of the invention, the adaptive filters may be utilized during normal operation by having them perform an echo cancellation operation on each of the signals that is to be supplied to the calibrated beamformer.

摘要翻译： 在具有扬声器和多个麦克风的免提通信环境中，波束形成器被校准用作声学回波消除器。 为了执行校准，与每个麦克风相对应地提供多个自适应滤波器，并且训练每个自适应滤波器以对由相应的一个麦克风所经历的环境的回波特性进行建模。 目标源被激活，从而产生由麦克风接收的声信号。 训练后的自适应滤波器然后用于通过例如使每个滤波器具有伪噪声信号来产生干扰信号。 然后将干扰信号中的相应信号与由麦克风提供的相应信号组合，从而产生组合信号。 然后组合信号用于使波束形成器适应以消除干扰信号。 在本发明的另一方面，自适应滤波器可以在正常操作期间通过使自适应滤波器对要提供给校准波束形成器的每个信号执行回波消除操作。

公开(公告)号：US20080004872A1

公开(公告)日：2008-01-03

申请号：US11574665

申请日：2005-09-07

申请人： Sven Nordholm ,  Kevin Fynn

发明人： Sven Nordholm ,  Kevin Fynn

IPC分类号： H04R1/40

CPC分类号： H04R1/1083 ,  A61F2011/145 ,  G10K11/178 ,  G10K2210/1081 ,  H04R1/1008 ,  H04R5/027 ,  H04R5/0335 ,  H04R2410/05 ,  H04R2430/20 ,  H04R2499/11

摘要： An apparatus for sound enhancement has at least two microphones (9) that provide a directional microphone array which is arranged to be pointed in the direction of a sound source. The directional microphone array thereby receives sound emitted by the sound source and generates sound signals. A processor (20) processors the sound signals generated by the microphone array to enhance the sound received by the directional microphone array from the sound source relative to other sound received by the directional microphone array. The processor (20) generates a corresponding enhanced signal (ES). Loud speakers (22) reproduce the enhanced signal as audible sound. Furthermore, sound suppression devices (7, 7a) are provided to suppress ambient should from reaching the eardrums of the user. This sound suppression acts in conjunction with the directional microphone array and the processor (20) which enhance the SOI to provide a listening environment in which the SOI is enhanced.

摘要翻译： 用于声音增强的装置具有至少两个麦克风（9），其提供被布置为指向声源的方向的定向麦克风阵列。 定向麦克风阵列由此接收由声源发出的声音并产生声音信号。 处理器（20）处理由麦克风阵列产生的声音信号，以增强由定向麦克风阵列从声源相对于由定向麦克风阵列接收的其它声音接收的声音。 处理器（20）产生相应的增强信号（ES）。 扬声器（22）将增强的信号再现为可听见的声音。 此外，提供声音抑制装置（7,7a）以抑制环境不能到达用户的耳膜。 该声音抑制与定向麦克风阵列和处理器（20）结合，该处理器（20）增强SOI以提供其中SOI被增强的收听环境。