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1. 发明授权

US09086475B2 Self-localization for a set of microphones 有权
标题翻译：一组麦克风的自定位
公开(公告)号：US09086475B2
公开(公告)日：2015-07-21
申请号：US13746333
申请日：2013-01-22
申请人： Willem Bastiaan Kleijn , Nikolay D. Gaubitch , Richard Heusdens
发明人： Willem Bastiaan Kleijn , Nikolay D. Gaubitch , Richard Heusdens
IPC分类号： G01S3/80 , G01S5/18 , G01S5/02
CPC分类号： G01S3/80 , G01S5/0278 , G01S5/0289 , G01S5/18
摘要： Provided are methods and systems for finding the location of sensors (e.g., microphones) with unknown internal delays based on a set of events (e.g., acoustic events) with unknown event time. A localization algorithm may iteratively run to compute the acoustic event times, the observation delays, and the relative locations of the events and the sensors.
摘要翻译：提供了用于基于具有未知事件时间的一组事件（例如，声学事件）来找到具有未知内部延迟的传感器（例如，麦克风）的位置的方法和系统。可以迭代地运行定位算法来计算声学事件时间，观测延迟以及事件和传感器的相对位置。

2. 发明申请

US20140204716A1 SELF-LOCALIZATION FOR A SET OF MICROPHONES 有权
标题翻译：一套麦克风的自定位
公开(公告)号：US20140204716A1
公开(公告)日：2014-07-24
申请号：US13746333
申请日：2013-01-22
申请人： Willem Bastiaan KLEIJN , Nikolay D. Gaubitch , Richard Heusdens
发明人： Willem Bastiaan KLEIJN , Nikolay D. Gaubitch , Richard Heusdens
IPC分类号： G01S3/80
CPC分类号： G01S3/80 , G01S5/0278 , G01S5/0289 , G01S5/18
摘要： Provided are methods and systems for finding the location of sensors (e.g., microphones) with unknown internal delays based on a set of events (e.g., acoustic events) with unknown event time. A localization algorithm may iteratively run to compute the acoustic event times, the observation delays, and the relative locations of the events and the sensors.
摘要翻译：提供了用于基于具有未知事件时间的一组事件（例如，声学事件）来找到具有未知内部延迟的传感器（例如，麦克风）的位置的方法和系统。可以迭代地运行定位算法来计算声学事件时间，观测延迟以及事件和传感器的相对位置。

3. 发明授权

US07120587B2 Sinusoidal model based coding of audio signals 失效
标题翻译：基于正弦模型的音频信号编码
公开(公告)号：US07120587B2
公开(公告)日：2006-10-10
申请号：US10169345
申请日：2001-10-31
申请人： Richard Heusdens , Renat Vafin , Willem Bastiaan Kleijn
发明人： Richard Heusdens , Renat Vafin , Willem Bastiaan Kleijn
IPC分类号： G10L19/02
CPC分类号： G10L19/02 , G10L21/0364 , G10L2019/0014
摘要： An apparatus and method of signal coding includes an analysis-by-synthesis algorithm for sinusoidal modeling. An input signal to be modeled is divided in time to produce a plurality of frames. Functions from a dictionary are selected to form an approximation of the section of the input signal contained in each frame, with the selection carried out based on a psychoacoustic norm. The function dictionary is made up of complex exponentials and these are selected iteratively to make up the section of the input signal contained in each frame. The psychoacoustic norm adapts after each iteration according to the changing masking threshold of the residual signal to be modeled in the next step.
摘要翻译：信号编码的装置和方法包括用于正弦建模的综合分析算法。要建模的输入信号被划分成时间以产生多个帧。选择来自字典的函数以形成包含在每个帧中的输入信号的部分的近似，其中基于心理声学范数进行选择。函数字典由复指数组成，并且这些被迭代地选择以构成包含在每个帧中的输入信号的部分。根据在下一步中建模的残差信号的改变屏蔽阈值，心理声学规范在每次迭代后适应。

4. 发明授权

US07020615B2 Method and apparatus for audio coding using transient relocation 失效
标题翻译：使用瞬态重定位的音频编码的方法和装置
公开(公告)号：US07020615B2
公开(公告)日：2006-03-28
申请号：US10003052
申请日：2001-11-02
申请人： Renat Vafin , Richard Heusdens , Steven Leonardus Josephus Dimphina Elisabeth Van De Par , Willem Bastiaan Kleijn
发明人： Renat Vafin , Richard Heusdens , Steven Leonardus Josephus Dimphina Elisabeth Van De Par , Willem Bastiaan Kleijn
IPC分类号： G10L19/00
CPC分类号： G10L19/02
摘要： An improved representation of transients in audio signals comprises modifying transient locations in such a way that a transient can occur only at a beginning of a sinusoidal segment. The modification procedure comprises the steps: detecting a beginning and an end of a transient using an energy-based approach with two sliding rectangular windows; moving samples between the beginning and the end of the transient to the locations specified by the segmentation used; and time-warping the signal parts in between the transients in order to fill the intervals between the modified transients.
摘要翻译：音频信号中的瞬变的改进的表示包括以这样的方式修改瞬态位置，使得瞬态可以仅在正弦段的开始处发生。修改过程包括以下步骤：使用具有两个滑动矩形窗口的基于能量的方法来检测瞬变的开始和结束; 将瞬态的开始和结束之间的样本移动到由所使用的分割指定的位置; 并且在瞬变之间对信号部分进行时间扭曲，以填补修改的瞬变之间的间隔。

5. 发明授权

US5812219A System and method for transmitting picture or sound signals 失效
标题翻译：用于传输图像或声音信号的系统和方法
公开(公告)号：US5812219A
公开(公告)日：1998-09-22
申请号：US551592
申请日：1995-11-01
申请人： Richard Heusdens
发明人： Richard Heusdens
IPC分类号： G06F17/10 , G06T9/00 , G10L19/00 , G10L19/02 , G10L19/032 , H03M7/30 , H04B1/66 , H04N1/41 , H04N7/00 , H04N19/60 , H04N5/14 , H04N7/12 , H04N9/64 , H04N11/02
CPC分类号： H04N19/527 , H04B1/667 , H04N19/86
摘要： Due to a reduction of block artefacts in the reconstructed signal, lapped signal transforms are more suitable than non-lapped transforms for compressing picture and sound signals. Nevertheless, the known lapped orthogonal transform is still accompanied by the artefacts. A system and method are described for transmitting picture or sound signals in which the artefacts remain absent also upon substantial compression and in which orthogonality and phase linearity of the transform is maintained. The transform matrix may be constructed from any arbitrary orthogonal K*K basis matrix C. It is indicated how the basis matrix can be obtained, for example from the DCT.
摘要翻译：由于重构信号中块伪像的减少，重叠信号变换比用于压缩图像和声音信号的非重叠变换更适合。然而，已知的重叠正交变换仍然伴随有人造物。描述了用于传输图像或声音信号的系统和方法，其中在大量压缩时，伪像仍然不存在，并且其中维持变换的正交性和相位线性。变换矩阵可以由任何正交的K * K基矩阵C构成。它表示如何可以例如从DCT获得基本矩阵。

6. 发明授权

US09064502B2 Speech intelligibility predictor and applications thereof 有权
标题翻译：语音清晰度预测及其应用
公开(公告)号：US09064502B2
公开(公告)日：2015-06-23
申请号：US13045303
申请日：2011-03-10
申请人： Cees H. Taal , Richard Hendriks , Richard Heusdens , Ulrik Kjems , Jesper Jensen
发明人： Cees H. Taal , Richard Hendriks , Richard Heusdens , Ulrik Kjems , Jesper Jensen
IPC分类号： G10L21/02 , G10L15/16 , G10L25/69
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经受处理算法和/或在噪声环境中被接收时，估计平均听众理解目标语音信号的能力。该应用还涉及一种改善听众对嘈杂环境中的目标语音信号的理解以及对应的设备单元的方法。本申请的目的是提供替代的目标可懂度度量，例如，适合在时频环境中使用的措施。本发明可以例如用于音频处理系统，例如收听系统，例如助听器系统。

7. 发明申请

US20100067710A1 NOISE SPECTRUM TRACKING IN NOISY ACOUSTICAL SIGNALS 有权
公开(公告)号：US20100067710A1
公开(公告)日：2010-03-18
申请号：US12550926
申请日：2009-08-31
申请人： Richard C. HENDRIKS , Jesper Jensen , Ulrik Kjems , Richard Heusdens
发明人： Richard C. HENDRIKS , Jesper Jensen , Ulrik Kjems , Richard Heusdens
IPC分类号： H04R29/00
CPC分类号： G10L21/0208 , G10L21/0216 , G10L2021/0575
摘要： The invention relates to a method of estimating noise power spectral density PSD in an input sound signal comprising a noise signal part and a target signal part. The invention further relates to a system to its use. The object of the present invention is to provide a scheme for estimating the noise PSD in an acoustic signal consisting of a target signal contaminated by acoustic noise. The problem is solved by a method comprising the steps of d) providing a digitized electrical input signal to a control path and performing; d1) storing a number of time frames of the input signal each comprising a predefined number N2 of digital time samples xn (n=1, 2, . . . , N2), corresponding to a frame length in time of L2=N2/fs; d2) performing a time to frequency transformation of the stored time frames on a frame by frame basis to provide corresponding spectra Y of frequency samples; d3) deriving a periodogram comprising the energy content |Y|2 for each frequency sample in a spectrum, the energy content being the energy of the sum of the noise and target signal; d4) applying a gain function G to each frequency sample of a spectrum, thereby estimating the noise energy level |Ŵ|2 in each frequency sample, |Ŵ|2=G·|Y|2; d5) dividing the spectra into a number Nsb2 of sub-bands, each sub-band comprising a predetermined number nsb2 of frequency samples, and assuming that the noise PSD level is constant across a sub-band; d6) providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band based on the non-zero noise energy levels of the frequency samples in the sub-band; d7) providing a second, improved estimate |Ñ|2 of the noise PSD level in a sub-band by applying a bias compensation factor B to the first estimate, |Ñ|2=B·|{circumflex over (N)}|2. The invention may e.g. be used in listening devices, e.g. hearing aids, mobile telephones, headsets, active earplugs, etc.

8. 发明授权

US08712074B2 Noise spectrum tracking in noisy acoustical signals 有权
标题翻译：噪声频谱跟踪噪声信号
公开(公告)号：US08712074B2
公开(公告)日：2014-04-29
申请号：US12550926
申请日：2009-08-31
申请人： Richard C. Hendriks , Jesper Jensen , Ulrik Kjems , Richard Heusdens
发明人： Richard C. Hendriks , Jesper Jensen , Ulrik Kjems , Richard Heusdens
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L21/0216 , G10L2021/0575
摘要： A method estimates noise power spectral density (PSD) in an input sound signal to generate an output for noise reduction of the input sound signal. The method includes storing frames of a digitized version of the input signal, each frame having a predefined number N2 of samples corresponding to a frame length in time of L2=N2/sampling frequency. It further includes performing a time to frequency transformation, deriving a periodogram comprising an energy content |Y|2 from the corresponding spectrum Y, applying a gain function G(k,m)=f(σs2(km),σw2l (k,m−1), |Y(k,m)|2), to estimate a noise energy level |Ŵ|2 in each frequency sample, where σs2 is the speech PSD and σw2 the noise PSD. It further includes dividing spectra into a number of sub-bands, and providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band and a second, improved estimate |{circumflex over (N)}|2 of the noise PSD level in a subband by applying a bias compensation factor B to the first estimate.
摘要翻译：方法估计输入声音信号中的噪声功率谱密度（PSD）以产生用于输入声音信号的噪声降低的输出。该方法包括存储输入信号的数字化版本的帧，每帧具有对应于L2 = N2 /采样频率的帧长度的预定数量的样本N2。它还包括执行时间到频率变换，从相应的频谱Y导出包括能量含量| Y | 2的周期图，应用增益函数G（k，m）= f（＆sgr; s2（km），＆sgr; w2l （k，m-1），| Y（k，m）| 2），以估计每个频率样本中的噪声能级|Ŵ| 2，其中＆sgr; s2是语音PSD，＆sgr; w2是噪声PSD。它还包括将频谱划分成多个子带，并且提供子带中的噪声PSD电平的第一估计|（在（N）} | 2）和第二估计|改善估计）} | 2，通过将偏置补偿因子B应用于第一估计，子带中的噪声PSD电平。

9. 发明申请

US20080097763A1 Combined Audio Coding Minimizing Perceptual Distortion 有权
标题翻译：组合音频编码最小化感知失真
公开(公告)号：US20080097763A1
公开(公告)日：2008-04-24
申请号：US11575149
申请日：2005-09-02
申请人： Steven Leonardus Josephus Van De Par , Nicolle Van Schijndel , Valery Kot , Richard Heusdens
发明人： Steven Leonardus Josephus Van De Par , Nicolle Van Schijndel , Valery Kot , Richard Heusdens
IPC分类号： G10L19/00
CPC分类号： G10L19/002 , G10L19/22
摘要： An audio encoder in which two or more preferably different encoders cooperate to generate a joint encoded audio signal. Encoding parameters of the two or more encoders are optimized in response to a measure of distortion of the joint encoded audio signal in accordance with a predetermined criterion. The distortion. measure is preferably a perceptual distortion measure. In one encoder embodiment comprising a sinusoidal and a waveform encoder, a constant total bit rate for each audio frame is distributed between the two encoders so as to minimize perceptual distortion for both the first and the second encoder. Other embodiments consider a set of encoding parameters that is larger than only those that minimize the perceptual distortion of the first encoder. In some embodiments, perceptual distortion may be minimized by optimizing encoding via optimizing entire encoding templates, i.e. a complex set of encoding parameters, for the separate encoders. The separate encoders may either be cascaded or operate in parallel, or in a combination of these. Two or more audio segments are preferably taken into account in the optimizing procedure. A corresponding audio decoder comprises separate decoders corresponding to the separate encoders of the audio encoder that encoded the audio signal. Decoded signal parts from these decoders are then added to produce the final audio signal. The presented audio encoding is efficient and provides a high sound quality because the encoding scheme is flexible and adapts to specific demands for each audio excerpt.
摘要翻译：一种音频编码器，其中两个或更多个优选地不同的编码器协作以产生联合编码的音频信号。响应于根据预定标准对联合编码音频信号的失真的测量，优化两个或更多个编码器的编码参数。失真。度量优选是感知失真度量。在包括正弦曲线和波形编码器的一个编码器实施例中，每个音频帧的恒定总比特率分布在两个编码器之间，以便最小化第一和第二编码器的感知失真。其他实施例考虑一组编码参数，其大于仅使最小化第一编码器的感知失真的编码参数。在一些实施例中，可以通过优化编码，通过优化整个编码模板（即编码参数的复杂集合）来分离编码器来最小化感知失真。单独的编码器可以级联或并联操作，也可以组合使用。在优化过程中优选考虑两个或多个音频段。对应的音频解码器包括对应于编码音频信号的音频编码器的分离编码器的单独的解码器。然后添加来自这些解码器的解码信号部分以产生最终音频信号。所呈现的音频编码是有效的并且提供高音质，因为编码方案是灵活的并且适应于每个音频片段的特定需求。

10. 发明申请

US20140126728A1 SYSTEM AND METHOD FOR EMITTING AND ESPECIALLY CONTROLLING AN AUDIO SIGNAL IN AN ENVIRONMENT USING AN OBJECTIVE INTELLIGIBILITY MEASURE 有权
标题翻译：使用目标智能测量在环境中发射和特别地控制音频信号的系统和方法
公开(公告)号：US20140126728A1
公开(公告)日：2014-05-08
申请号：US14116995
申请日：2011-05-11
申请人： Hans Van Der Schaar , Oosterom Han , Richard Heusdens , Richard Hendriks
发明人： Hans Van Der Schaar , Oosterom Han , Richard Heusdens , Richard Hendriks
IPC分类号： G10L21/02 , H04R29/00
CPC分类号： G10L21/02 , G08B3/10 , H04R29/00 , H04R29/007 , H04R2227/009
摘要： Public address systems or other systems for emitting audio signals, like music, speech or announcements, in different locations like supermarkets, schools, universities, auditoriums are widely known. These systems usually comprise an audio source, for example a microphone or a recorder, and a plurality of loudspeakers, which are locally distributed in the locations, for emitting the audio signal from the audio source. The invention proposes a system (1) and a method for emitting an audio signal (2, 8) in an environment (3), the system (1) comprising: an audio source for providing the audio signal (2, 8), at least one loudspeaker (4) for emitting the audio signal (2), at least one microphone (5) for receiving an acoustic signal (6) from the environment (3), whereby the acoustic signal (6) is based on the audio signal (2) and may comprise disturbing components (7), and with an analyzing module (13) for analyzing the acoustic signal (6) and for providing an intelligibility measure from an objective intelligibility measure method, whereby the intelligibility measure is used as a feedback signal.
摘要翻译：在诸如超市，学校，大学，礼堂等不同地点发布音频信号，如音乐，演讲或公告的公共广播系统或其他系统是众所周知的。这些系统通常包括音频源，例如麦克风或记录器，以及多个扬声器，其本地分布在该位置，用于从音频源发出音频信号。本发明提出了一种在环境（3）中发出音频信号（2,8）的系统（1）和方法，所述系统（1）包括：用于提供音频信号（2,8）的音频源，用于发射音频信号的至少一个扬声器（4），用于从环境（3）接收声信号（6）的至少一个麦克风（5），由此声信号（6）基于音频信号（2），并且可以包括干扰组件（7），以及用于分析声信号（6）的分析模块（13），并且用于从目标可懂度测量方法提供可懂度度量度，由此将可理解度量度用作反馈信号。

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