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31. 发明申请

WO2016169604A1 AN AUDIO SIGNAL PROCESSING APPARATUS FOR PROCESSING AN INPUT EARPIECE AUDIO SIGNAL UPON THE BASIS OF A MICROPHONE AUDIO SIGNAL 审中-公开
标题翻译：用于处理基于麦克风音频信号的输入音频信号的音频信号处理装置
公开(公告)号：WO2016169604A1
公开(公告)日：2016-10-27
申请号：PCT/EP2015/058809
申请日：2015-04-23
申请人： HUAWEI TECHNOLOGIES CO., LTD. , FALLER, Christof , FAVROT, Alexis , GROSCHE, Peter , LANG, Yue
发明人： FALLER, Christof , FAVROT, Alexis , GROSCHE, Peter , LANG, Yue
IPC分类号： G10L21/034
CPC分类号： G10L21/034 , G10L21/0205 , G10L25/78 , G10L25/84
摘要： The invention relates to an audio signal processing apparatus (100) for processing an input earpiece audio signal (x) upon the basis of a microphone audio signal (y), the audio signal processing apparatus (100) comprising a voice activity detector (101) being configured to determine a voice activity indicator signal (x vad ) upon the basis of the input earpiece audio signal (x), a noise magnitude determiner (103) being configured to determine a microphone noise magnitude indicator signal (wy) upon the basis of the microphone audio signal (y), a gain factor determiner (105) being configured to determine a gain factor signal (Δ G ) upon the basis of the voice activity indicator signal (x vad ) and the microphone noise magnitude indicator signal (w y ), and a weighter (107) being configured to weight the input earpiece audio signal (x) by the gain factor signal (Δ G ) to obtain an output earpiece audio signal.
摘要翻译：本发明涉及一种用于根据麦克风音频信号（y）处理输入听筒音频信号（x）的音频信号处理设备（100），音频信号处理设备（100）包括语音活动检测器（101）被配置为基于输入听筒音频信号（x）来确定语音活动指示符信号（xvad），噪声量值确定器（103）被配置为基于所述语音活动指示符信号麦克风音频信号（y），增益因子确定器（105）被配置为基于语音活动指示符信号（xvad）和麦克风噪声幅度指示符信号（wy）来确定增益因子信号（ΔG），以及重心（107）被配置为通过增益因子信号（ΔG）对输入听筒音频信号（x）进行加权，以获得输出听筒音频信号。

32. 发明申请

WO2016096021A1 SURROUND SOUND RECORDING FOR MOBILE DEVICES 审中-公开
标题翻译：移动设备的环绕声录音
公开(公告)号：WO2016096021A1
公开(公告)日：2016-06-23
申请号：PCT/EP2014/078558
申请日：2014-12-18
申请人： HUAWEI TECHNOLOGIES CO., LTD. , FALLER, Christof , FAVROT, Alexis , GROSCHE, Peter , LANG, Yue
发明人： FALLER, Christof , FAVROT, Alexis , GROSCHE, Peter , LANG, Yue
IPC分类号： H04R3/00 , H04R1/32 , H04R1/40
CPC分类号： H04R5/027 , H04R1/326 , H04R1/406 , H04R3/005 , H04R5/04 , H04R2430/21 , H04R2499/11 , H04S3/00 , H04S2400/15
摘要： The present invention is directed to a microphone arrangement (100) and a method (900) using the microphone arrangement (100) for recording surround sound in a mobile device (200). The microphone arrangement (100) comprises a first and a second microphone (102) and (103) arranged at a first distance (d 1 ) to each other and configured to obtain a stereo signal, and comprises a third microphone (103) configured to obtain a steering signal (DOA, 1-DOA) together with at least one of the first and second microphone (102) and (103) and/or with a fourth microphone (104). The microphone arrangement (100) also comprises a processor (105) configured to separate the stereo signal into a front stereo signal (FL, FR) and a back stereo signal (BL, BR) based on the steering signal (DOA, 1-DOA).
摘要翻译：本发明涉及一种在移动设备（200）中使用麦克风装置（100）来记录环绕声的麦克风装置（100）和方法（900）。麦克风装置（100）包括第一和第二麦克风（102）和（103），它们以彼此之间的第一距离（d1）布置并被配置为获得立体声信号，并且包括被配置为获得的第三麦克风（103）与第一和第二麦克风（102）和（103）中的至少一个和/或与第四麦克风（104）一起的转向信号（DOA，1-DOA）。麦克风装置（100）还包括处理器（105），其被配置为基于转向信号（DOA，1-DOA）将立体声信号分离成前立体声信号（FL，FR）和后立体声信号（BL，BR））。

33. 发明申请

WO2013029225A1 PARAMETRIC MULTICHANNEL ENCODER AND DECODER 审中-公开
标题翻译：参数多通道编码器和解码器
公开(公告)号：WO2013029225A1
公开(公告)日：2013-03-07
申请号：PCT/CN2011/079051
申请日：2011-08-29
申请人： HUAWEI TECHNOLOGIES CO., LTD. , FALLER, Christof , VIRETTE, David , LANG, Yue
发明人： FALLER, Christof , VIRETTE, David , LANG, Yue
IPC分类号： G10L19/00
CPC分类号： G10L19/008
摘要： The present invention relates to a parametric multichannel encoder 501 for encoding a multichannel signal comprising a first channel signal and a second channel signal, the parametric multichannel encoder 501 comprising an estimator 505 for estimating an interaural time difference between the first channel signal and the second channel signal to obtain an estimate of the interaural time difference, the estimator 505 being further configured to determine a fuzziness indicator, the fuzziness indicator indicating a grade of non-reliability of the estimate of the interaural time difference; a downmix signal generator 507 for generating a downmix signal from the first channel signal and the second channel signal; and a multiplexer 511 for multiplexing the downmix signal, the interaural time difference, and the fuzziness parameter to obtain an encoded signal.
摘要翻译：本发明涉及用于对包括第一信道信号和第二信道信号的多信道信号进行编码的参数化多信道编码器501，参数多信道编码器501包括用于估计第一信道信号和第二信道之间的耳间时间差的估计器505 信号以获得昼间时间差的估计，估计器505还被配置为确定模糊指示符，模糊指示符指示玄度时间差的估计的不可靠度等级; 降混信号发生器507，用于从第一通道信号和第二通道信号产生下混合信号; 以及用于多路复用下混合信号，耳间时间差和模糊参数以获得编码信号的多路复用器511。

34. 发明申请

WO2012055687A1 ECHO SUPPRESSION COMPRISING MODELING OF LATE REVERBERATION COMPONENTS 审中-公开
标题翻译：包含最新版本组件建模的ECHO抑制
公开(公告)号：WO2012055687A1
公开(公告)日：2012-05-03
申请号：PCT/EP2011/067486
申请日：2011-10-06
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , KUECH, Fabian , SCHMIDT, Markus , FALLER, Christof , FAVROT, Alexis
发明人： KUECH, Fabian , SCHMIDT, Markus , FALLER, Christof , FAVROT, Alexis
IPC分类号： G10L21/02 , H04M9/08
CPC分类号： H04R3/002 , G10L2021/02082 , H04M9/085
摘要： An apparatus (200) for computing filter coefficients (H[k,m]) for an adaptive filter (210) is disclosed. The adaptive filter is used for filtering a microphone signal (140, 430) so as to suppress an echo due to a loudspeaker signal (130, 420). The apparatus (200) comprises: an echo decay modeling means (465) for modeling a decay behavior of an acoustic environment (120) and for providing a corresponding echo decay parameter (τ; α m ); and computing means (270; 370) for computing the filter coefficients (H[k,m]) of the adaptive filter (210) on the basis of the echo decay parameter (τ; α m ). A corresponding method comprises: providing echo decay parameters (τ; Ĝ[k,m]; α m ) determined by means of an echo decay modeling means (465); and computing the filter coefficients (H[k,m]) of the adaptive filter (210) on the basis of the echo decay parameters (τ; α m ).
摘要翻译：公开了一种用于计算自适应滤波器（210）的滤波器系数（H [k，m]）的装置（200）。自适应滤波器用于对麦克风信号（140,430）进行滤波，以便抑制由扬声器信号（130,420）引起的回波。所述装置（200）包括：回波衰减建模装置（465），用于对声学环境（120）的衰减行为进行建模并提供对应的回波衰减参数（t; am）; 以及用于基于回波衰减参数（t; am）计算自适应滤波器（210）的滤波器系数（H [k，m]）的计算装置（270; 370）。相应的方法包括：提供通过回波衰减建模装置（465）确定的回波衰减参数（t; G [k，m]; am）; 并根据回波衰减参数（t; am）计算自适应滤波器（210）的滤波器系数（H [k，m]）。

35. 发明申请

WO2009095161A1 APPARATUS AND METHOD FOR COMPUTING FILTER COEFFICIENTS FOR ECHO SUPPRESSION 审中-公开
标题翻译：用于计算ECHO抑制的过滤系数的装置和方法
公开(公告)号：WO2009095161A1
公开(公告)日：2009-08-06
申请号：PCT/EP2009/000270
申请日：2009-01-16
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , KUECH, Fabian , KALLINGER, Markus , FALLER, Christof , FAVROT, Alexis
发明人： KUECH, Fabian , KALLINGER, Markus , FALLER, Christof , FAVROT, Alexis
IPC分类号： G10L21/02 , H04B3/23 , H04M9/08
CPC分类号： H04M9/082 , G10L21/0208 , G10L2021/02082 , H04B3/237
摘要： An embodiment of an apparatus (200) for computing filter coefficients for an adaptive filter (210) for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes extraction means (250) for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and computing means (270) for computing the filter coefficients for the adaptive filter (210) on the basis of the extracted stationary component signal or the extracted non-stationary component signal.
摘要翻译：用于计算用于滤波麦克风信号以便抑制由于扬声器信号引起的回波的自适应滤波器（210）的滤波器系数的装置（200）的实施例包括用于提取静止分量信号或非平滑分量信号的提取装置（250）来自扬声器信号的静止分量信号或来自扬声器信号的信号，以及用于基于所提取的固定分量信号或提取的非均衡信号计算自适应滤波器（210）的滤波器系数的计算装置（270）固定分量信号。

36. 发明申请

WO2006111370A1 A METHOD AND DEVICE FOR REMOVING ECHO IN A MULTI-CHANNEL AUDIO SIGNAL 审中-公开
标题翻译：用于在多声道音频信号中移除ECHO的方法和装置
公开(公告)号：WO2006111370A1
公开(公告)日：2006-10-26
申请号：PCT/EP2006/003582
申请日：2006-04-19
申请人： EPFL (ECOLE POLYTECHNIQUE FEDERALE DE LAUSANNE) , FALLER, Christof
发明人： FALLER, Christof
IPC分类号： H04M9/08
CPC分类号： H03G9/005 , H04M9/082
摘要： Acoustic echo control and noise suppression is an important part of any "handsfree" telecommunication system, such as telephony or audio or video conferencing systems. Bandwidth and computational complexity constraints have prevented that stereo or multi-channel telecommunication systems have been widely applied. The advantages are very low complexity, high robustness, scalability to multi-channel audio without a need for loudspeaker signal distortion, and efficient integration of echo and noise control in the same algorithm. The proposed method of processing multi-channels audio loudspeakers signals and at least one microphone signal, comprises the steps of: transforming the input microphone signals (yl (n), y2 (n), ....., yM (n)) into input microphone short-time spectra, computing a combined loudspeaker signal short-time spectrum [X(i,k)] from the loudspeaker signals, (xl (n), x2 (n), ...., xL (n)), computing a combined microphone signal short-time spectrum [Y(i,k)] from the input microphone signal, (yl (n), y2 (n), ….., yM (n)), estimating a magnitude or power spectrum of the echo in the combined microphone signal short-time spectrum, computing a gain filter (G(i,k)) for magnitude modification of the input microphone short-time spectra, applying the gain filter to at least one of the input microphone spectra, converting the filtered input microphone spectra into the time domain (el (n), e2 (n) , .…., eM (n)).
摘要翻译：声学回声控制和噪声抑制是任何“免提”电信系统的重要部分，例如电话或音频或视频会议系统。带宽和计算复杂度约束阻止了立体声或多声道电信系统已被广泛应用。其优点是非常低的复杂性，高鲁棒性，多通道音频的可扩展性，而不需要扬声器信号失真，并且在相同的算法中有效地集成回波和噪声控制。所提出的处理多声道音频扬声器信号和至少一个麦克风信号的方法包括以下步骤：将输入麦克风信号（yl（n），y2（n），...，yM（n））变换为从扬声器信号（xl（n），x2（n），...，xL（n））中计算组合的扬声器信号短时频谱[X（i，k）]，），从输入麦克风信号（yl（n），y2（n），...，yM（n））计算组合的麦克风信号短时频谱[Y（i，k）]，估计幅度或功率在组合的麦克风信号短时间频谱中的回波频谱，计算用于输入麦克风短时频谱的幅度修正的增益滤波器（G（i，k）），将增益滤波器应用于至少一个输入麦克风将滤波后的输入麦克风频谱转换成时域（el（n），e2（n），...，eM（n））。

37. 发明申请

WO2006072270A1 COMPACT SIDE INFORMATION FOR PARAMETRIC CODING OF SPATIAL AUDIO 审中-公开
标题翻译：空间音频参考编码的紧凑信息
公开(公告)号：WO2006072270A1
公开(公告)日：2006-07-13
申请号：PCT/EP2005/010595
申请日：2005-09-30
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , AGERE SYSTEMS INC. , HERRE, Jürgen , FALLER, Christof
发明人： HERRE, Jürgen , FALLER, Christof
IPC分类号： G10L19/00 , H04S3/00
CPC分类号： G10L19/008
摘要： At an audio encoder, cue codes are generated for one or more audio channels, wherein a combined cue code (e.g., a combined inter-channel correlation (ICC) code) is generated by combining two or more estimated cue codes, each estimated cue code estimated from a group of two or more channels. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels. Received cue codes include a combined cue code (e.g., a combined ICC code). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein two or more derived cue codes are derived from the combined cue code, and each derived cue code is applied to generate two or more synthesized channels.
摘要翻译：在音频编码器处，为一个或多个音频通道生成提示码，其中通过组合两个或多个估计的提示码来生成组合的提示码（例如，组合的信道间相关（ICC）码），每个估计的提示码从一组两个或多个渠道估计。在音频解码器处，E个发送的音频信道被解码以产生C个回放音频信道。接收的提示码包括组合的提示码（例如，组合的ICC码）。一个或多个传输的信道被混合以产生一个或多个上混频道。通过将提示码应用于一个或多个上混通道来合成一个或多个回放通道，其中从组合提示码导出两个或多个派生提示码，并且应用每个派生的提示码以生成两个或多个合成通道。

38. 发明申请

WO2006060279A1 PARAMETRIC CODING OF SPATIAL AUDIO WITH OBJECT-BASED SIDE INFORMATION 审中-公开
标题翻译：具有基于对象的信息的空间音频参数编码
公开(公告)号：WO2006060279A1
公开(公告)日：2006-06-08
申请号：PCT/US2005/042772
申请日：2005-11-22
申请人： AGERE SYSTEMS INC. , FALLER, Christof
发明人： FALLER, Christof
IPC分类号： G10L19/00
CPC分类号： H04S1/002 , G10L19/008
摘要： A binaural cue coding scheme involving one or more object-based cue codes, wherein an object-based cue code directly represents a characteristic of an auditory scene corresponding to the audio channels, where the characteristic is independent of number and positions of loudspeakers used to create the auditory scene. Examples of object-based cue codes include the angle of an auditory event, the width of the auditory event, the degree of envelopment of the auditory scene, and the directionality of the auditory scene.
摘要翻译：一种涉及一个或多个基于对象的提示码的双耳提示编码方案，其中基于对象的提示码直接表示对应于音频信道的听觉场景的特征，其中该特征独立于用于创建的扬声器的数量和位置听觉场面。基于对象的提示代码的示例包括听觉事件的角度，听觉事件的宽度，听觉场景的包络程度以及听觉场景的方向性。

39. 发明申请

WO2005094125A1 FREQUENCY-BASED CODING OF AUDIO CHANNELS IN PARAMETRIC MULTI-CHANNEL CODING SYSTEMS 审中-公开
标题翻译：参数多通道编码系统中频率通道的基于频率的编码
公开(公告)号：WO2005094125A1
公开(公告)日：2005-10-06
申请号：PCT/US2005/005605
申请日：2005-02-23
申请人： AGERE SYSTEMS INC. , FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FALLER, Christof , HERRE, Juergen
发明人： FALLER, Christof , HERRE, Juergen
IPC分类号： H04S3/00
CPC分类号： H04S3/00 , G10L19/008 , H04S2420/03
摘要： For a multi-channel audio signal, parametric coding is applied to different subsets of audio input channels for different frequency regions. For example, for a 5.1 surround sound signal having five regular channels and one low-frequency (LFE) channel, binaural cue coding (BCC) can be applied to all six audio channels for sub-bands at or below a specified cut-off frequency, but to only five audio channels (excluding the LFE channel) for sub-bands above the cut-off frequency. Such frequency-based coding of channels can reduce the encoding and decoding processing loads and/or size of the encoded audio bitstream relative to parametric coding techniques that are applied to all input channels over the entire frequency range.
摘要翻译：对于多声道音频信号，参数编码被应用于不同频率区域的音频输入通道的不同子集。例如，对于具有五个常规频道和一个低频（LFE）频道的5.1环绕声信号，可以将双耳提示编码（BCC）应用于等于或低于指定截止频率的子频带的所有六个音频通道，但是对于截止频率以上的子频带，只有五个音频通道（不包括LFE通道）。信道的这种基于频率的编码可以相对于在整个频率范围上应用于所有输入信道的参数编码技术来减少编码和解码处理负载和/或编码音频比特流的大小。

40. 发明申请

WO2005069274A1 APPARATUS AND METHOD FOR CONSTRUCTING A MULTI-CHANNEL OUTPUT SIGNAL OR FOR GENERATING A DOWNMIX SIGNAL 审中-公开
标题翻译：用于构造多通道输出信号或产生下行信号的装置和方法
公开(公告)号：WO2005069274A1
公开(公告)日：2005-07-28
申请号：PCT/EP2005/000408
申请日：2005-01-17
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , AGERE SYSTEMS INC. , HERRE, Jürgen , FALLER, Christof
发明人： HERRE, Jürgen , FALLER, Christof
IPC分类号： G10L19/00
CPC分类号： G10L19/008 , H04S3/02 , H04S2420/03
摘要： The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing (324) first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating (322) a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.
摘要翻译：用于使用输入信号和参数侧信息构造多通道输出信号的装置，包括从原始多通道信号导出的第一输入通道和第二输入通道的输入信号以及描述通道之间的相互关系的参数侧信息多通道原始信号使用基本通道来合成（324）彼此不同的假定收听者位置一侧的第一和第二输出声道。由于一致性测量，基本通道彼此不同。通过通过输入通道的组合计算（322）这些通道中的一个通道的基本通道来减小基本通道（例如左和左环绕重构通道）之间的相干性，该组合由相干性测量确定。因此，由于近似的原始前/后相干性，可以获得重建的高主观质量。

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