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公开(公告)号：US10433056B2
公开(公告)日：2019-10-01
申请号：US16197696
申请日：2018-11-21
Applicant: Huawei Technologies Co., Ltd.
Inventor： Christof Faller , Alexis Favrot , Peter Grosche , Martin Pollow , Jürgen Geiger
Abstract: An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).
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公开(公告)号：US10375503B2
公开(公告)日：2019-08-06
申请号：US15786278
申请日：2017-10-17
Applicant: Huawei Technologies Co., Ltd.
Inventor： Michael Buerger , Heinrich Löllmann , Walter Kellermann , Peter Grosche , Yue Lang
Abstract: A wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the apparatus includes a sound field synthesizer for generating sound field drive signals for causing the array of loudspeakers to generate one or more sound fields at one or more audio zones, a binaural renderer for generating binaural drive signals for causing the array of loud-speakers to generate specified sound pressures at at least two positions, wherein the at least two positions are determined based on a detected position and/or orientation of a listener, and a decision unit for deciding whether to generate the drive signals using the sound field synthesizer or using the binaural renderer.
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公开(公告)号：US10199032B2
公开(公告)日：2019-02-05
申请号：US15952864
申请日：2018-04-13
Applicant: Huawei Technologies Co., Ltd.
Inventor： Wenyu Jin , Peter Grosche
IPC: G10K11/178 , G10L21/0232 , G10L21/0208
Abstract: A signal processor for determining a plurality of drive signals for driving a plurality of loudspeakers to cancel a reverberation effect in a listening area, wherein the signal processor is configured to determine from one or more measured audio signals a plurality of measured physical coefficients in a basis of physical sound functions, such that a sum of the physical sound functions, weighted with the plurality of measured physical coefficients approximates the one or more measured audio signals, wherein at least half of the plurality of measured physical coefficients are zero, determine a residual error between the plurality of measured physical coefficients and a plurality of desired physical coefficients, estimate a transfer function describing a transformation from the plurality of desired physical coefficients to the plurality of measured physical coefficients, based on the determined residual error, and update the plurality of drive signals based on the estimated transfer function.
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公开(公告)号：US10178491B2
公开(公告)日：2019-01-08
申请号：US15411859
申请日：2017-01-20
Applicant: Huawei Technologies Co., Ltd.
Inventor： Christof Faller , Alexis Favrot , Liyun Pang , Peter Grosche , Yue Lang
Abstract: The disclosure relates to an apparatus for manipulating an input audio signal associated to a spatial audio source within a spatial audio scenario, wherein the spatial audio source has a certain distance to a listener within the spatial audio scenario. The apparatus comprises an exciter adapted to manipulate the input audio signal to obtain an output audio signal, and a controller adapted to control parameters of the exciter for manipulating the input audio signal based on the certain distance.
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公开(公告)号：US20180262832A1
公开(公告)日：2018-09-13
申请号：US15978688
申请日：2018-05-14
Applicant: Huawei Technologies Co., Ltd.
Inventor： Peter Grosche , Karim Helwani , Christian Schörkhuber , Franz Zotter , Robert Höldrich , Matthias Frank
CPC classification number: H04R3/005 , G10L25/21 , H03G3/20 , H03G3/3005 , H04R1/406 , H04R3/04 , H04R2430/01
Abstract: A sound signal processing apparatus including a plurality of microphones, where each microphone is configured to receive the sound signal from the target source, a processor configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone of the microphones and a second power measure on the basis of the sound signal from the target source received by at least a second microphone of the microphones, which is located more distant from the target source than the first microphone, and the processor is further configured to determine a gain factor on the basis of a ratio between the second power measure and the first power measure, and an amplifier configured to apply the gain factor to the sound signal from the target source received by the first microphone.
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Patent Agency Ranking
公开(公告)号：US09985597B2
公开(公告)日：2018-05-29
申请号：US15082081
申请日：2016-03-28
Applicant: Huawei Technologies Co., Ltd.
Inventor： Peter Grosche , Yue Lang , Qing Zhang
Abstract: A digital compressor for compressing an input audio signal is presented. The digital compressor comprises a compression gain control for providing a compression gain parameter, and a compression parameter determiner for determining a compression ratio from the compression gain parameter. The compression parameter determiner may be configured to weight the compression gain parameter by a predetermined weight factor to obtain the compression ratio. The digital compressor further comprises an auxiliary signal generator for manipulating the input audio signal in dependence of the compression ratio to obtain a first auxiliary signal, and a combiner unit for combining the first auxiliary signal with the compression gain parameter to obtain a second auxiliary signal, and for combining the input audio signal with the second auxiliary signal to obtain the compressed audio signal.
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公开(公告)号：US20170078821A1
公开(公告)日：2017-03-16
申请号：US15362275
申请日：2016-11-28
Applicant: Huawei Technologies Co., Ltd.
Inventor： Christof Faller , Alexis Favrot , Yue Lang , Peter Grosche
CPC classification number: H04S7/306 , H04S1/002 , H04S3/008 , H04S2420/01
Abstract: The disclosure is based on the finding that acoustic near-field transfer functions indicating acoustic near-field propagation channels between loudspeakers and ears of a listener can be employed to pre-process audio signals. Therefore, acoustic near-field distortions of the audio signals can be mitigated. The pre-processed audio signals can be presented to the listener using a wearable frame, wherein the wearable frame comprises the loudspeakers for audio presentation. The disclosure can allow for a high quality rendering of audio signals as well as a high listening comfort for the listener. The disclosure can provide the following advantages. By means of a loudspeaker selection as a function of a spatial audio source direction, cues related to the listener's ears can be generated, making the approach more robust with regard to front/back confusion. The approach can further be extended to an arbitrary number of loudspeaker pairs.
Abstract translation: 本公开是基于以下发现：可以采用指示收听者的扬声器和耳朵之间的声学近场传播信道的声学近场传递函数来预处理音频信号。 因此，可以减轻音频信号的声学近场失真。 预处理的音频信号可以使用可穿戴的框架呈现给收听者，其中可穿戴的框架包括用于音频呈现的扬声器。 本公开可以允许音频信号的高质量渲染以及听众的高听觉舒适度。 本公开可以提供以下优点。 通过扬声器选择作为空间音频源方向的功能，可以产生与听众耳朵相关的线索，使得该方法在前/后混淆方面更加健壮。 该方法可以进一步扩展到任意数量的扬声器对。
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公开(公告)号：US20160344356A1
公开(公告)日：2016-11-24
申请号：US15225553
申请日：2016-08-01
Applicant: Huawei Technologies Co., Ltd.
Inventor： Peter Grosche , Yue Lang , Qing Zhang
Abstract: An audio compression system for compressing an input audio signal, and the audio compression system comprises a digital filter for filtering the input audio signal, where the digital filter comprises a frequency transfer function having a magnitude over frequency, where the magnitude is formed by an equal loudness curve of a human ear to obtain a filtered audio signal, and a compressor which is configured to compress the input audio signal upon the basis of the filtered audio signal to obtain a compressed audio signal.
Abstract translation: 一种用于压缩输入音频信号的音频压缩系统，并且音频压缩系统包括用于对输入音频信号进行滤波的数字滤波器，其中数字滤波器包括频率幅度超过频率的频率传递函数，其中幅度由等于 获得滤波后的音频信号的人耳的响度曲线;以及压缩器，被配置为基于经滤波的音频信号来压缩输入音频信号，以获得压缩音频信号。
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29.发明申请Apparatus and Method for Compressing a Set of N Binaural Room Impulse Responses 审中-公开压缩一组N双耳室冲击响应的装置和方法公开(公告)号：US20160212564A1
公开(公告)日：2016-07-21
申请号：US15082087
申请日：2016-03-28
Applicant: Huawei Technologies Co., Ltd.
Inventor： Simone Fontana , Karim Helwani , Peter Grosche
IPC: H04S7/00 , G10L19/008
CPC classification number: H04S7/306 , G10L19/008 , H04S3/004 , H04S2400/01 , H04S2400/03 , H04S2420/01 , H04S2420/07
Abstract: An apparatus and a method for compressing a set of N binaural room impulse responses, BRIR, wherein each channel of an N channel audio signal is convolved with the corresponding compressed set of N BRIR. The apparatus may comprise at least one analyzing and compressor module adapted to separate an input binaural room impulse response signal into a first binaural signal set provided to the binauralization processing of the initial part of the BRIR (early part) and a second binaural signal set provided to the binauralization processing of the final part of the BRIR (late part) via a downmix module; a binauralization module adapted to obtain a binaural signal based on convolving the N channel audio signal with the first binaural signal set and the second binaural signal set.
Abstract translation: 一种用于压缩一组N个双耳室脉冲响应BRIR的装置和方法，其中N个声道音频信号的每个声道与N个BRIR的对应压缩集进行卷积。 该装置可以包括至少一个分析和压缩器模块，其适于将输入的双耳室脉冲响应信号分离成提供给BRIR的初始部分（早期部分）的双耳化处理的第一双耳信号组和提供的第二双耳信号组 通过下混模块对BRIR（后期部分）最后部分的双耳化处理; 一种双耳化模块，其适于基于将N声道音频信号与第一双耳信号组和第二双耳信号集合进行卷积来获得双耳信号。
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公开(公告)号：US10284993B2
公开(公告)日：2019-05-07
申请号：US15722637
申请日：2017-10-02
Inventor： Michael Buerger , Heinrich Löllmann , Walter Kellermann , Karim Helwani , Peter Grosche , Yue Lang
Abstract: A local wave field synthesis apparatus, which includes a determination module for determining desired sound pressures and desired particle velocity vectors at a plurality of control points, a computation module for computing sound pressures and particle velocity vectors at the plurality of control points based on a set of filter parameters, an optimization module for computing an optimum set of filter parameters by jointly optimizing computed sound pressures towards the desired sound pressures and computed particle velocity vectors towards the desired particle velocity vectors, and a generator module for generating the drive signals based on the optimum set of filter parameters, wherein the plurality of control points are located on one or more contours around the one or more audio zones.
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