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1. 发明申请

WO2008102207A3 METHOD OF ESTIMATING NOISE LEVELS IN A COMMUNICATION SYSTEM 审中-公开
标题翻译：通信系统中噪声水平的估算方法
公开(公告)号：WO2008102207A3
公开(公告)日：2008-10-30
申请号：PCT/IB2007004498
申请日：2007-12-20
申请人： SKYPE LTD , VOS KOEN , SORENSEN KARSTEN VANDBORG , BERGENHEIM JON
发明人： VOS KOEN , SORENSEN KARSTEN VANDBORG , BERGENHEIM JON
IPC分类号： G10L21/02 , G10L21/0208 , G10L21/0232
CPC分类号： G10L21/0208 , G10L21/0232
摘要： A method of estimating noise in data containing voice information and noise comprising the steps of receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.
摘要翻译：一种估计包含语音信息和噪声的数据中的噪声的方法，包括以下步骤：接收作为输入值序列的数据; 通过对输入值应用第一非线性映射来变换数据，其中随着输入值在幅度平滑变换的数据中映射的导数函数在幅度上减小; 以及通过应用与第一非线性映射相反的第二非线性映射来变换平滑的变换数据以确定输入数据中的噪声的估计。

2. 发明申请

WO2014019596A3 PROCESSING AUDIO SIGNALS 审中-公开
标题翻译：处理音频信号
公开(公告)号：WO2014019596A3
公开(公告)日：2014-04-10
申请号：PCT/EP2012059937
申请日：2012-05-28
申请人： SKYPE , SORENSEN KARSTEN VANDBORG
发明人： SORENSEN KARSTEN VANDBORG
IPC分类号： H04L12/18 , H04R3/00
CPC分类号： H03G3/3005 , H03G3/3089 , H04L12/1827 , H04R3/005 , H04R2499/11
摘要： A method, user device and computer program product for processing audio signals during a communication session between a user device and a remote node. The method comprising: receiving a plurality of audio signals at audio input means at the user device including at least one primary audio signal and unwanted signals; receiving direction of arrival information of the audio signals at a gain control means; providing to the gain control means known direction of arrival information representative of at least some of said unwanted signals; processing the audio signals at the gain control means by applying a level of gain to generate a gain controlled signal for transmission to the remote node, wherein the level of gain applied is dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information.
摘要翻译：一种用于在用户设备和远程节点之间的通信会话期间处理音频信号的方法，用户设备和计算机程序产品。该方法包括：在用户设备处的音频输入装置处接收包括至少一个主音频信号和不需要的信号的多个音频信号; 在增益控制装置处接收音频信号的到达方向信息; 向增益控制装置提供代表至少一些所述不想要的信号的已知的到达方向信息; 通过施加增益水平来处理增益控制装置处的音频信号以生成增益控制信号以发送到远程节点，其中所施加的增益水平取决于音频信号的到达方向信息与已知的到达方向信息。

3. 发明申请

WO2012110614A4 PROCESSING AUDIO SIGNALS 审中-公开
标题翻译：处理音频信号
公开(公告)号：WO2012110614A4
公开(公告)日：2012-11-08
申请号：PCT/EP2012052718
申请日：2012-02-16
申请人： SKYPE , SORENSEN KARSTEN VANDBORG , PENA JESUS DE VICENTE
发明人： SORENSEN KARSTEN VANDBORG , PENA JESUS DE VICENTE
IPC分类号： G10L21/0208 , H04M9/08
CPC分类号： G10L21/0208
摘要： A method of reducing noise in a signal received at a processing stage of an acoustic system, the method comprising, at the processing stage: identifying at least one frequency at which a system gain of the acoustic system is above an average system gain of the acoustic system; providing a noise attenuation factor for reducing noise in the signal for the at least one frequency, the noise attenuation factor for the at least one frequency based on the system gain for that frequency; and applying the noise attenuation factor to a component of the signal at that frequency.
摘要翻译：一种降低在声学系统的处理阶段接收的信号中的噪声的方法，所述方法包括：在处理阶段：识别声学系统的系统增益高于声学系统的平均系统增益的至少一个频率系统; 提供用于减少所述至少一个频率的所述信号中的噪声的噪声衰减因子，所述噪声衰减因子是基于所述频率的系统增益的所述至少一个频率的噪声衰减因子; 以及将噪声衰减因子应用于该频率处的信号分量。

4. 发明专利

GB2495472A Device including an acoustic beamformer and means processing received audio signals for echo suppression 未知
公开(公告)号：GB2495472A
公开(公告)日：2013-04-17
申请号：GB201116869
申请日：2011-09-30
申请人： SKYPE
发明人： SORENSEN KARSTEN VANDBORG
IPC分类号： H04R3/00 , G10K11/34 , H04R1/40
摘要： The device (e.g. a mobile telephone) comprises means for outputting audio signals of a communication event. The device queries a data store to obtain an indication of an echo direction, wherein the echo direction relates to a direction from which audio signals output from the audio output means are likely to be received at a microphone array. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event, wherein the beamformer coefficients are determined such that the beamformer applies echo suppression to audio signals received at the microphone array from the indicated echo direction.

5. 发明专利

GB2495129A Selecting beamformer coefficients using a regularization signal with a delay profile matching that of an interfering signal 未知
公开(公告)号：GB2495129A
公开(公告)日：2013-04-03
申请号：GB201116846
申请日：2011-09-30
申请人： SKYPE
发明人： SORENSEN KARSTEN VANDBORG , AHGREN PER
IPC分类号： H04R3/00 , G10K11/34
摘要： A plurality of sensors, such as microphones, receive a signal from an interfering source (e.g. 206, Fig. 1). Each sensor produces an electrical signal (x1, x2, x3) upon receipt of the interfering signal. The electrical signals are produced at respective different delay times, creating an interference delay pattern. A plurality of regularization signals (n1, n2, n3) are generating, having a delay pattern matching the interference delay pattern. In a preferred embodiment, each of the regularization signals is a white noise signal. The regularization signal is perceived as emanating from the source of the interference. The regularization signals are used in determining beamformer coefficients for use in detecting a target signal (e.g. d1, Fig. 1). For example, the regularization signals may be mixed with signals from respective sensors, before the signals are input to a processing block 404 for determining beamformer coefficients. Alternatively, the beamformer coefficients may be based solely on the generated regularization signals. The invention may be applied, in a microphone array embodiment, to detect audio signals suffering from localised interference. However RF communication, radar and sonar embodiments are also briefly mentioned.

6. 发明专利

GB2491173A Setting gain applied to an audio signal based on direction of arrival (DOA) information 未知
公开(公告)号：GB2491173A
公开(公告)日：2012-11-28
申请号：GB201108885
申请日：2011-05-26
申请人： SKYPE
发明人： SORENSEN KARSTEN VANDBORG
IPC分类号： H03G3/32 , G10K11/34 , H03G3/30
摘要： A method of processing audio signals is disclosed in which the operation of a gain control unit 228 is influenced on a frame-by-frame basis by direction of arrival information 406 produced from a plurality of microphones 402. The method comprises: receiving a plurality of audio signals at the audio input of a user device, the signals including at least one primary audio signal and unwanted signals; receiving direction of arrival information of the audio signals at a gain control means; providing to the gain control means known direction of arrival information representative of at least some of said unwanted signals; processing the audio signals at the gain control means by applying a level of gain to generate a gain controlled signal for transmission to a remote node, wherein the level of gain applied is dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information. The method prevents the automatic gain control (AGC) 228 from adjusting the gain applied to a frame (fig.6) when unwanted signals arrive at the microphone assembly 208 during a call.

7. 发明专利

GB2466670B Speech encoding 未知
公开(公告)号：GB2466670B
公开(公告)日：2012-11-14
申请号：GB0900140
申请日：2009-01-06
申请人： SKYPE
发明人： VOS KOEN , SORENSEN KARSTEN VANDBORG , JENSEN SOREN SKAK
IPC分类号： G10L19/00 , G10L19/02 , G10L19/06 , G10L19/07 , G10L25/24
摘要： A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.

8. 发明专利

GB2476041A Soft transition between the audio bandwidth of a signal before and after a switch in sampling rate 未知
公开(公告)号：GB2476041A
公开(公告)日：2011-06-15
申请号：GB0921462
申请日：2009-12-08
申请人： SKYPE LTD
发明人： STROMMER STEFAN , SORENSEN KARSTEN VANDBORG , JENSEN SOREN SKAK , VOS KOEN , BERGENHEIM JON
IPC分类号： G10L19/24 , G10L19/26 , G10L21/02
摘要： A method and apparatus for transmitting an audio signal over a communication channel comprising encoding the audio signal with an encoder 204 using a first sampling rate, filtering the audio signal using a first cut off frequency, the first cut off frequency being chosen in dependence upon the first sampling rate, and transmitting the encoded and filtered audio signal over the communication channel. The presence of a condition in which the sampling rate of the encoder 204 is to be switched to a second sampling rate at a switching time is determined and if the condition has been determined to be present, the cut off frequency used in the filtering step is gradually changed from the first cut off frequency to a second cut off frequency, the second cut off frequency being chosen in dependence upon the second sampling rate, such that the audio bandwidth of the transmitted signal changes gradually when the sampling rate is switched to the second sampling rate.

9. 发明专利

GB2497343B Processing audio signals 未知
公开(公告)号：GB2497343B
公开(公告)日：2014-11-26
申请号：GB201121147
申请日：2011-12-08
申请人： SKYPE
发明人： SORENSEN KARSTEN VANDBORG
IPC分类号： H04R3/02 , G10K11/34 , H04R1/40 , H04R3/12

10. 发明专利

GB2496660A Processing received audio signals with a beamformer and an echo canceller 未知
公开(公告)号：GB2496660A
公开(公告)日：2013-05-22
申请号：GB201119932
申请日：2011-11-18
申请人： SKYPE
发明人： SORENSEN KARSTEN VANDBORG
IPC分类号： H04R3/00 , H04R1/40
摘要： The processing method comprises receiving audio signals at a plurality of microphones of the device, processing at least one of the audio signals received by the plurality of microphones to generate a first characteristic, a beamformer appliesg beamformer coefficients to the received audio signals, thereby generating a beamformer output, processing the beamformer output to generate a second characteristic, applying echo cancelling means to the beamformer output, thereby suppressing an echo resulting from audio signals output from the audio output means from the beamformer output. An operating parameter of the echo cancelling means is determined using a relationship between the first and second characteristics.

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