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：EP2474170B1

公开(公告)日：2017-05-31

申请号：EP09782555.8

申请日：2009-09-03

申请人： Robert Bosch GmbH

发明人： SMAAK, Marc ,  JANSE, C.P. ,  TCHANG, Chen ,  VAN STUIVENBERG, L.C.A.

IPC分类号： H04R3/00

CPC分类号： H04R27/00 ,  H04M11/06 ,  H04R2227/009 ,  H04S2420/05

公开(公告)号：EP3113183A1

公开(公告)日：2017-01-04

申请号：EP15755932.9

申请日：2015-02-12

申请人： National Institute of Information and Communications Technology

发明人： SHIGA, Yoshinori

IPC分类号： G10L21/007 ,  G10L21/0208

CPC分类号： G10L21/0332 ,  G10L13/033 ,  G10L19/0204 ,  G10L21/007 ,  G10L21/0208 ,  G10L21/0232 ,  G10L21/0364 ,  G10L25/15 ,  G10L2021/02165 ,  H04R27/00 ,  H04R2227/009

摘要： [Object] To provide a speech intelligibility improving apparatus capable of generating highly intelligible speech in various environments without unnecessarily amplifying sound volume.
[Solution] A speech intelligibility improving apparatus 250 includes: an envelope surface extracting unit 292 extracting, from a spectrum of speech signal 254 as an object of processing, a curve representing a general outline of peaks of spectral envelope in contact with or along local peaks of spectral envelope of the spectrum; a noise adapting unit 300 modifying spectrum of speech signal 254 based on the curve extracted by envelope surface extracting unit 292; and a sinusoidal wave speech synthesizing unit 305 generating a modified speech signal 260 for the speech improved in intelligibility based on the spectrum modified by noise adapting unit 300.

摘要翻译： 提供能够在各种环境中产生高度可理解的语音的语音清晰度提高装置，而不必不必要地放大音量。 语音清晰度提高装置250包括：包络表面提取单元292，从作为处理对象的语音信号254的频谱中提取表示与局部峰值接触或沿着局部峰值的频谱包络的​​峰值的总体轮廓的曲线 的光谱包络谱; 噪声适应单元300，基于由包络面提取单元292提取的曲线来修改语音信号254的频谱; 以及正弦波语音合成单元305，其基于由噪声适应单元300修改的频谱，生成用于可懂度改善的语音的修改语音信号260。

公开(公告)号：EP2814266A1

公开(公告)日：2014-12-17

申请号：EP13171814.0

申请日：2013-06-13

申请人： Airbus Operations GmbH

发明人： Scheel, Henning

IPC分类号： H04R29/00 ,  G10L21/02 ,  E04B1/99

CPC分类号： H04R29/007 ,  H04R2227/009

摘要： A method for optimizing the speech intelligibility in a passenger compartment of a vehicle comprises establishing an acoustic room model of the passenger compartment, establishing background noise data inside and outside the compartment, establishing speaker data, determining a speech intelligibility factor on the basis of the acoustic room model, the background noise data and the speaker data at at least one first test point in the compartment; comparing the determined speech intelligibility factor with a predetermined speech intelligibility factor and modifying at least one of the speaker characteristics, the geometrical layout and acoustically active material of at least one installation in the compartment in case the predetermined speech intelligibility factor exceeds the determined speech intelligibility factor. Thereby, physical tests may be minimized and optimal speech intelligibility can be ensured.

摘要翻译： 一种用于在车辆的乘客室优化语音清晰度方法包括乘客舱的声学室模型的确立之，建立背景噪声数据内和车厢外，建立扬声器数据，确定性的挖掘一个语音可懂度因子的声的基础上 室模型，背景噪声数据以及在所述隔室的至少一个第一测试点的扬声器数据; 确定性开采语音可懂度因子与预定的语音可懂度因子进行比较和修改的扬声器特性的至少一个，所述几何布局以及至少一个安装的声学活性物质的情况下，隔室中的预定的语音可懂度因子超过所述确定性开采语音可懂度因子 ， 由此，物理测试可以被最小化和最佳语音可懂度能得到保证。

公开(公告)号：EP1219134A4

公开(公告)日：2007-08-01

申请号：EP00965639

申请日：2000-09-20

申请人： MASS ENTPR PTY LTD

发明人： GROOTHUIS HENRICUS JOHANNES FR

IPC分类号： H04R3/14

CPC分类号： H04R3/14 ,  H04R27/00 ,  H04R2227/009

摘要： A method of treating an audio signal is disclosed, comprising introducing a phase shift to the audio signal, the magnitude of the phase shift being dependent upon the frequency of the audio signal. A signal control system implementing the method is also disclosed, described for an active loudspeaker having at least one sound transducer, each sound transducer having an amplifier connected directly thereto, the signal control system provided between a signal source and the amplifier and comprising crossover means arranged to deliver a frequency limited portion of the signal to the amplifier; and phase shift means arranged to introduce a frequency dependent phase shift to the signal prior to being input to the amplifier, wherein the magnitude of the phase shift is dependent upon the frequency of the audio signal. The method and system increases the perceived clarify of the sound by the listener.

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：EP3255900A1

公开(公告)日：2017-12-13

申请号：EP17174435.2

申请日：2017-06-05

申请人： Sonos Inc.

发明人： SHIH, Shao-Fu

IPC分类号： H04R3/00 ,  H04R27/00 ,  H04W24/02 ,  G10L21/0216

CPC分类号： G10L21/0216 ,  G06F3/162 ,  G10L21/0208 ,  G10L21/0264 ,  G10L2021/02082 ,  G10L2021/02166 ,  H04R3/005 ,  H04R27/00 ,  H04R2227/003 ,  H04R2227/005 ,  H04R2227/009 ,  H04W24/02

摘要： A set of signal measures is received, wherein each signal measure in the set of signal measures corresponds to a respective audio signal received by a device in a media playback system which is processed based on a first set of audio processing algorithms. A plurality of signal measures is identified in the set of signal measures. Audio signals corresponding to the identified plurality of signal measures are processed by one or more devices in the media playback system to improve a signal measure of each of the audio signals. The audio signals are processed based on a second set of audio processing algorithms. The processed audio signals are then combined into a combined audio signal.

摘要翻译： 接收一组信号测量，其中该组信号测量中的每个信号测量对应于由媒体回放系统中的设备接收的相应音频信号，其基于第一组音频处理算法进行处理。 在该组信号测量中识别多个信号测量。 对应于所识别的多个信号测量的音频信号由媒体回放系统中的一个或多个设备处理，以改善每个音频信号的信号测量。 基于第二组音频处理算法处理音频信号。 经处理的音频信号然后被组合成组合的音频信号。

公开(公告)号：EP3230977A1

公开(公告)日：2017-10-18

申请号：EP15791196.7

申请日：2015-10-29

申请人： Qualcomm Incorporated

发明人： KIM, Lae-Hoon ,  MOHAMMAD, Asif Iqbal ,  VISSER, Erik

IPC分类号： G10K11/178 ,  G10L21/02 ,  H04R3/02 ,  H04R5/02 ,  H04S7/00

CPC分类号： G10K11/178 ,  G10L21/02 ,  G10L2021/02082 ,  H04M9/082 ,  H04R3/02 ,  H04R5/02 ,  H04R27/00 ,  H04R2227/009 ,  H04R2499/13 ,  H04S7/00 ,  H04S7/302

摘要： A crosstalk cancelation technique reduces feedback in a shared acoustic space by canceling out some or all parts of sound signals that would otherwise be produced by a loudspeaker to only be captured by a microphone that, recursively, would cause these sounds signals to be reproduced again on the loudspeaker as feedback. Crosstalk cancelation can be used in a multichannel acoustic system (MAS) comprising an arrangement of microphones, loudspeakers, and a processor to together enhance conversational speech between in a shared acoustic space. To achieve crosstalk cancelation, a processor analyzes the inputs of each microphone, compares it to the output of far loudspeaker(s) relative to each such microphone, and cancels out any portion of a sound signal received by the microphone that matches signals that were just produced by the far loudspeaker(s) and sending only the remaining sound signal (if any) to such far loudspeakers.

摘要翻译： 串扰消除技术通过消除声音信号的一些或全部部分来减少共享声学空间中的反馈，所述声音信号否则将由扬声器产生，以便仅由麦克风捕获，所述麦克风递归地将导致这些声音信号再次被再现 扬声器作为反馈。 串音消除可用于包括麦克风，扬声器和处理器在内的多声道声学系统（MAS），以共同增强共享声学空间之间的对话语音。 为了实现串扰消除，处理器分析每个麦克风的输入，将其与每个这样的麦克风的远扬声器的输出进行比较，并且消除由麦克风接收到的匹配信号的声音信号的任何部分 由远端扬声器产生并且仅将剩余的声音信号（如果有的话）发送到这样的远端扬声器。

公开(公告)号：EP2842123A1

公开(公告)日：2015-03-04

申请号：EP12723791.5

申请日：2012-05-16

申请人： Nuance Communications, Inc.

发明人： BUCK, Markus ,  HAULICK, Tim ,  MATHEJA, Timo

IPC分类号： G10L15/00 ,  G10L15/22

CPC分类号： G10L21/0205 ,  G10L15/00 ,  G10L15/32 ,  G10L21/0272 ,  G10L25/48 ,  G10L2021/02166 ,  H04M1/6075 ,  H04M3/568 ,  H04R3/005 ,  H04R3/12 ,  H04R27/00 ,  H04R2227/009 ,  H04R2499/13

摘要： A multi-mode speech communication system is described that has different operating modes for different speech applications. A speech service compartment contains multiple system users, multiple input microphones that develop microphone input signals from the system users to the system, and multiple output loudspeakers that develop loudspeaker output signals from the system to the system users. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes the microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application. The output processing module processes application output communications from the speech applications to produce loudspeaker output signals to the system users, wherein for each different speech application, the loudspeaker output signals are directed only to system users currently active in that speech application. The signal processing module dynamically controls the processing of the microphone input signals and the loudspeaker output signals to respond to changes in currently active system users for each application.

摘要翻译： 描述了具有用于不同语音应用的不同操作模式的多模式语音通信系统。 信号处理模块与语音应用通信，并且包括输入处理模块和输出处理模块。 输入处理模块处理麦克风输入信号以产生用于每个语音应用的集合用户输入信号，该用户输入信号仅限于该语音应用的当前活动系统用户。 输出处理模块处理来自语音应用的应用输出通信，以向系统用户产生扬声器输出信号，其中对于每个不同的语音应用，扬声器输出信号仅被引导到当前在该语音应用中有效的系统用户。

公开(公告)号：EP2088590B1

公开(公告)日：2014-12-10

申请号：EP07790221.1

申请日：2007-06-26

申请人： Sony Computer Entertainment Inc.

发明人： YAMASHITA, Kosei ,  HONDA, Shinichi

IPC分类号： G10L21/04 ,  H04R5/04

CPC分类号： G10L19/008 ,  G10L19/0204 ,  H04R5/04 ,  H04R2227/009 ,  H04R2420/01 ,  H04S2420/07