公开(公告)号：EP3255634A1

公开(公告)日：2017-12-13

申请号：EP17173455.1

申请日：2017-05-30

申请人： Oticon A/S

发明人： JENSEN, Jesper ,  KJEMS, Ulrik ,  BERTELSEN, Andreas Thelander ,  PEDERSEN, Michael Syskind

IPC分类号： G10L21/0264

CPC分类号： G10L21/0232 ,  A61F11/06 ,  G10L21/0264 ,  H04B1/3833 ,  H04M1/6041 ,  H04R1/1083 ,  H04R25/50 ,  H04R25/505

摘要： The application relates to an audio processing device comprising a) at least one input unit for providing time-frequency representation Y(k,n) of an electric input signal representing a time variant sound signal consisting of target speech signal components S(k,n) from a target sound source TS and noise signal components N(k,n) from other sources than the target sound source, where k and n are frequency band and time frame indices, respectively, b) a noise detection and/or reduction system configured to b1) determine an a posteriori signal to noise ratio estimate γ(k,n) of said electric input signal, and to b2) determine an a priori target signal to noise signal ratio estimate ζ(k,n) of said electric input signal from said a posteriori signal to noise ratio estimate γ(k,n) based on a recursive decision directed algorithm. The a priori target signal to noise signal ratio estimate ζ(k,n) for the n th timeframe is determined from the a priori target signal to noise signal ratio estimate ζ(k,n-1) for the (n-1) th timeframe and the a posteriori signal to noise ratio estimate γ(k,n) for the n th timeframe. The application further relates to a method of of estimating an a priori signal to noise ratio. Thereby improved noise reduction may be provided. The invention may e.g. be used for the hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

摘要翻译： 本申请涉及一种音频处理设备，包括a）至少一个输入单元，用于提供表示由目标语音信号分量S（k，n）组成的时变声音信号的电输入信号的时频表示Y（k，n） ）和来自除目标声源之外的其他源的噪声信号分量N（k，n），其中k和n分别是频带和时间帧索引，b）噪声检测和/或降低系统 被配置为b1）确定所述电输入信号的后验信噪比估计值γ（k，n），并且b2）确定所述电输入信号的先验目标信噪比估计值ζ（k，n） 基于递归决策导向算法从所述后验信噪比估计γ（k，n）中检测信号。 根据第（n-1）个时间帧的先验目标信噪比估计值ζ（k，n-1）确定第n个时间帧的先验目标信噪比估计值ζ（k，n） 和第n个时间帧的后验信噪比估计γ（k，n）。 本申请还涉及一种估计先验信噪比的方法。 由此可以提供改进的降噪。 本发明可以例如 用于助听器，耳机，耳机，有源耳机保护系统，免提电话系统，移动电话等。

公开(公告)号：EP1785007B1

公开(公告)日：2013-11-20

申请号：EP05774069.8

申请日：2005-08-22

申请人： Oticon A/S

发明人： RASMUSSEN, Karsten Bo ,  PETERSEN, Kim Spetzler ,  KJEMS, Ulrik

IPC分类号： H04R29/00

CPC分类号： H04R29/006

摘要： The invention relates to a communication device having at least two microphones, where in order to match the microphone performance in respect of the phase response a correction filter in the form of a IIR filter is implemented and where the amplitude of the transfer function for the correction filter is the inverse of the difference between the two microphone amplitudes.

公开(公告)号：EP3190587A1

公开(公告)日：2017-07-12

申请号：EP16193246.2

申请日：2012-08-24

申请人： Retune DSP ApS ,  Oticon A/S

发明人： KJEMS, Ulrik ,  JENSEN, Jesper

IPC分类号： G10L21/0232 ,  H04M9/08 ,  H04R3/00 ,  G10L21/0216 ,  H04R1/10 ,  H04R3/02

CPC分类号： G10L15/20 ,  G10L21/0232 ,  G10L2021/02166 ,  H04M9/082 ,  H04R1/1083 ,  H04R3/005 ,  H04R3/02 ,  H04R2225/43 ,  H04R2227/009

摘要： The application relates to a method for audio signal processing. The application further relates to a method of processing signals obtained from a multi-microphone system. The object of the present application is to reduce undesired noise sources and residual echo signals from an initial echo cancellation step. The problem is solved by
receiving M communication signals in frequency subbands where M is at least two;
processing the M subband communication signals in each subband with a blocking matrix ( 203,303,403 ) of M rows and N linearly independent columns in each subband, where N >=1 and N , to obtain N target-cancelled signals in each subband;
processing the M subband communication signals and the N target-cancelled signals in each subband with a set of beamformer coefficients ( 204,304,404 ) to obtain a beamformer output signal in each subband;
processing the communication signals with a target absence detector ( 309 ) to obtain a target absence signal in each subband;
using the target absence signal to obtain an inverse target-cancelled covariance matrix of order N ( 310,410 ) in each band;
processing the N target-cancelled signals in each subband with the inverse target-cancelled covariance matrix in a quadratic form ( 312, 412 ) to yield a real-valued noise correction factor in each subband;
using the target absence signal to obtain an initial estimate ( 311, 411 ) of the noise power in the beamformer output signal averaged over recent frames with target absence in each subband;
multiplying the initial noise estimate with the noise correction factor to obtain a refined estimate ( 417 ) of the power of the beamformer output noise signal component in each subband;
processing the refined estimate of the power of the beamformer output noise signal component with the magnitude of the beamformer output to obtain a postfilter gain value in each subband;
processing the beamformer output signal with the postfilter gain value ( 206,306,406 ) to obtain a postfilter output signal in each subband;
processing the postfilter output subband signals through a synthesis filterbank ( 207,307,407 ) to obtain an enhanced beamformed output signal where the target signal is enhanced by attenuation of noise signal components. This has the advantage of providing improved sound quality and reduction of undesired signal components such as the late reverberant part of an acoustic echo signal. The invention may e.g. be used for headsets, hearing aids, active ear protection systems, mobile telephones, teleconferencing systems, karaoke systems, public address systems, mobile communication devices, hands-free communication devices, voice control systems, car audio systems, navigation systems, audio capture, video cameras, and video telephony.

摘要翻译： 本申请涉及用于音频信号处理的方法。 本申请还涉及一种处理从多麦克风系统获得的信号的方法。 本申请的目的是从初始回声消除步骤减少不希望的噪声源和残余回声信号。 该问题通过在频率子带中接收M个通信信号来解决，其中M至少为2; 在每个子带中用M行和N个线性独立列的分块矩阵（203,303,403）处理每个子带中的M个子带通信信号，其中N≥1且N

公开(公告)号：EP3190587B1

公开(公告)日：2018-10-17

申请号：EP16193246.2

申请日：2012-08-24

申请人： Oticon A/S ,  Retune DSP ApS

发明人： KJEMS, Ulrik ,  JENSEN, Jesper

IPC分类号： G10L21/0232 ,  H04M9/08 ,  H04R3/00 ,  G10L21/0216 ,  H04R1/10 ,  H04R3/02

CPC分类号： G10L15/20 ,  G10L21/0232 ,  G10L2021/02166 ,  H04M9/082 ,  H04R1/1083 ,  H04R3/005 ,  H04R3/02 ,  H04R2225/43 ,  H04R2227/009

摘要： The application relates to a method for audio signal processing. The application further relates to a method of processing signals obtained from a multi-microphone system. The object of the present application is to reduce undesired noise sources and residual echo signals from an initial echo cancellation step. The problem is solved by receiving M communication signals in frequency subbands where M is at least two; processing the M subband communication signals in each subband with a blocking matrix ( 203,303,403 ) of M rows and N linearly independent columns in each subband, where N >=1 and N

公开(公告)号：EP1785007A1

公开(公告)日：2007-05-16

申请号：EP05774069.8

申请日：2005-08-22

申请人： Oticon A/S

发明人： RASMUSSEN, Karsten Bo ,  PETERSEN, Kim Spetzler ,  KJEMS, Ulrik

IPC分类号： H04R3/00 ,  H04R3/04

CPC分类号： H04R29/006

摘要： The invention relates to a communication device having at least two microphones, where in order to match the microphone performance in respect of the phase response a correction filter in the form of a IIR filter is implemented and where the amplitude of the transfer function for the correction filter is the inverse of the difference between the two microphone amplitudes.

摘要翻译： 本发明涉及具有至少两个麦克风的通信设备，其中为了匹配相对于相位响应的麦克风性能，实现了IIR滤波器形式的校正滤波器，并且其中用于校正的传递函数的幅度 滤波器是两个麦克风幅度之差的倒数。

公开(公告)号：EP3462452A1

公开(公告)日：2019-04-03

申请号：EP18200053.9

申请日：2012-08-24

申请人： Oticon A/S

发明人： KJEMS, Ulrik ,  JENSEN, Jesper

IPC分类号： G10L21/0232 ,  H04M9/08 ,  H04R3/00 ,  G10L21/0216 ,  H04R1/10 ,  H04R3/02

摘要： The application relates to a method for audio signal processing. The application further relates to a method of processing signals obtained from a multi-microphone system. The object of the present application is to reduce undesired noise sources and residual echo signals from an initial echo cancellation step. The problem is solved by
receiving M communication signals in frequency subbands where M is at least two;
processing the M subband communication signals in each subband with a blocking matrix (203,303,403) of M rows and N linearly independent columns in each subband, where N >=1 and N to obtain N target-cancelled signals in each subband;
processing the M subband communication signals and the N target-cancelled signals in each subband with a set of beamformer coefficients (204,304,404) to obtain a beamformer output signal in each subband;
processing the communication signals with a target absence detector (309) to obtain a target absence signal in each subband;
using the target absence signal to obtain an inverse target-cancelled covariance matrix of order N (310,410) in each band;
processing the N target-cancelled signals in each subband with the inverse target-cancelled covariance matrix in a quadratic form (312, 412) to yield a real-valued noise correction factor in each subband;
using the target absence signal to obtain an initial estimate (311, 411 ) of the noise power in the beamformer output signal averaged over recent frames with target absence in each subband;
multiplying the initial noise estimate with the noise correction factor to obtain a refined estimate (417) of the power of the beamformer output noise signal component in each subband;
processing the refined estimate of the power of the beamformer output noise signal component with the magnitude of the beamformer output to obtain a postfilter gain value in each subband;
processing the beamformer output signal with the postfilter gain value (206,306,406) to obtain a postfilter output signal in each subband;
processing the postfilter output subband signals through a synthesis filterbank (207,307,407) to obtain an enhanced beamformed output signal where the target signal is enhanced by attenuation of noise signal components. This has the advantage of providing improved sound quality and reduction of undesired signal components such as the late reverberant part of an acoustic echo signal. The invention may e.g. be used for headsets, hearing aids, active ear protection systems, mobile telephones, teleconferencing systems, karaoke systems, public address systems, mobile communication devices, hands-free communication devices, voice control systems, car audio systems, navigation systems, audio capture, video cameras, and video telephony.

公开(公告)号：EP3252766A1

公开(公告)日：2017-12-06

申请号：EP16171986.9

申请日：2016-05-30

申请人： Oticon A/S

发明人： JENSEN, Jesper ,  KJEMS, Ulrik ,  BERTELSEN, Andreas Thelander ,  PEDERSEN, Michael Syskind

IPC分类号： G10L21/0264

CPC分类号： G10L21/0232 ,  A61F11/06 ,  G10L21/0264 ,  H04B1/3833 ,  H04M1/6041 ,  H04R1/1083 ,  H04R25/50 ,  H04R25/505

摘要： The application relates to an audio processing device comprising a) at least one input unit for providing time-frequency representation Y(k,n) of an electric input signal representing a time variant sound signal consisting of target speech signal components S(k,n) from a target sound source TS and noise signal components N(k,n) from other sources than the target sound source, where k and n are frequency band and time frame indices, respectively, b) a noise detection and/or reduction system configured to b1) determine an a posteriori signal to noise ratio estimate γ(k,n) of said electric input signal, and to b2) determine an a priori target signal to noise signal ratio estimate ζ(k,n) of said electric input signal from said a posteriori signal to noise ratio estimate γ(k,n) based on a recursive decision directed algorithm. The a priori target signal to noise signal ratio estimate ζ(k,n) for the n th timeframe is determined from the a priori target signal to noise signal ratio estimate ζ(k,n-1) for the (n-1) th timeframe and the a posteriori signal to noise ratio estimate γ(k,n) for the n th timeframe. The application further relates to a method of of estimating an a priori signal to noise ratio. Thereby improved noise reduction may be provided. The invention may e.g. be used for the hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

公开(公告)号：EP2372700A1

公开(公告)日：2011-10-05

申请号：EP10156220.5

申请日：2010-03-11

申请人： Oticon A/S

发明人： TAAL, Cees H. ,  HENDRIKS, Richard ,  HEUSDENS, Richard ,  KJEMS, Ulrik ,  JENSEN, Jesper

IPC分类号： G10L19/00

CPC分类号： G10L25/69

摘要： The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.

摘要翻译： 本申请涉及一种提供语音清晰度预测值的方法，用于当所述目标语音信号受到处理算法和/或在嘈杂环境中接收时估计平均收听者理解目标语音信号的能力。 本申请还涉及一种改善嘈杂环境中的收听者对目标语音信号的理解的方法以及相应的设备单元。 本申请的目的是提供一种替代的客观可理解性度量，例如， 这是一种适用于时频环境的措施。 本发明可以例如 用于音频处理系统，例如 听音系统，例如 助听器系统。