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公开(公告)号:US20110029308A1
公开(公告)日:2011-02-03
申请号:US12813350
申请日:2010-06-10
申请人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
发明人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
IPC分类号: G10L15/00
CPC分类号: G10L25/78
摘要: The present invention relates to means and methods of classifying speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.
摘要翻译: 本发明涉及在语音通信系统,设备,电话和方法中分类语音和音乐信号的方法和方法,更具体地说,涉及当通过通信链路检测到语音或音乐时自动化控制的系统,设备和方法 。 本发明提供了一种用于监视音频信号,分析所选择的音频信号分量,将分析结果与预定阈值进行比较并将音频信号分类为语音或音乐的新型系统和方法。
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公开(公告)号:US20100082339A1
公开(公告)日:2010-04-01
申请号:US12567787
申请日:2009-09-27
申请人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich , Hariharan Ganapathy Kathirvelu
发明人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich , Hariharan Ganapathy Kathirvelu
IPC分类号: G10L21/02
CPC分类号: G10L21/0208
摘要: By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and/or cancellation protocols at the appropriate time via analog and/or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.
摘要翻译: 通过监视蜂窝电话正在操作的位置中的风噪声,以及通过模拟和/或数字信号处理在合适的时间应用降噪和/或消除协议,可以显着地减少进入通信的风噪声 系统。
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公开(公告)号:US08457320B2
公开(公告)日:2013-06-04
申请号:US12833804
申请日:2010-07-09
申请人: Alon Konchitsky , Alberto D Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
发明人: Alon Konchitsky , Alberto D Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
IPC分类号: H04R29/00
CPC分类号: H04R3/007
摘要: A special purpose machine measures and modulates communication signals that are parsed into frames. Frames of signals modulated and measured to have certain qualities are deemed to be the result of wind noise. Frames of wind noise are cancelled from further use within a communication system.
摘要翻译: 专用机器测量和调制被分析成帧的通信信号。 被调制和测量以具有某些品质的信号帧被认为是风噪声的结果。 在通信系统中进一步使用风噪声的帧被取消。
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公开(公告)号:US08340964B2
公开(公告)日:2012-12-25
申请号:US12813350
申请日:2010-06-10
申请人: Alon Konchitsky , Alberto D Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
发明人: Alon Konchitsky , Alberto D Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
CPC分类号: G10L25/78
摘要: The present invention relates to means and methods of classifying speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.
摘要翻译: 本发明涉及在语音通信系统,设备,电话和方法中分类语音和音乐信号的方法和方法,更具体地说,涉及当通过通信链路检测到语音或音乐时自动化控制的系统,设备和方法 。 本发明提供了一种用于监视音频信号,分析所选择的音频信号分量,将分析结果与预定阈值进行比较并将音频信号分类为语音或音乐的新型系统和方法。
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公开(公告)号:US08433564B2
公开(公告)日:2013-04-30
申请号:US12795188
申请日:2010-06-07
申请人: Alon Konchitsky , Alberto D Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
发明人: Alon Konchitsky , Alberto D Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
CPC分类号: G10L21/0208
摘要: A noisy signal is picked up by a microphone, digitized by an Analog to Digital Converter and fed to a processor for analysis and wind noise reduction. Most of noise reduction methods are based on the assumption that the interfering noise is stationary or slowly varying compared with speech. This assumption allows “learning” the characteristics of the noise between speech pauses and, based on a noise estimate, to build different filters that reduce the noise. In the case of wind noise this basic assumption is not valid. Wind noise is highly non-stationary, its power and spectral characteristics vary greatly. Because wind noise is not stationary, regular noise reduction methods cannot be used to reduce wind noise. For reducing wind noise effects in a device, the presence of wind should be detected reliably and then a novel approach presented here must be applied to eliminate the wind noise.
摘要翻译: 噪声信号由麦克风拾取,由模数转换器数字化,并馈送到处理器进行分析和减少风噪声。 大多数降噪方法是基于干扰噪声与语音相比是静止的或缓慢变化的假设。 该假设允许“学习”语音暂停之间的噪声的特性,并且基于噪声估计来建立减少噪声的不同滤波器。 在风噪声的情况下,这个基本假设是无效的。 风噪声非常不稳定,其功率和光谱特性差异很大。 由于风噪声不稳定,因此不能采用常规的降噪方法来减少风噪声。 为了减少设备中的风噪声影响,应该可靠地检测风的存在,然后必须采用这里提出的新颖方法来消除风噪声。
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公开(公告)号:US20110007906A1
公开(公告)日:2011-01-13
申请号:US12833804
申请日:2010-07-09
申请人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
发明人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
IPC分类号: G10K11/16
CPC分类号: H04R3/007
摘要: A special purpose machine measures and modulates communication signals that are parsed into frames. Frames of signals modulated and measured to have certain qualities are deemed to be the result of wind noise. Frames of wind noise are cancelled from further use within a communication system.
摘要翻译: 专用机器测量和调制被分析成帧的通信信号。 被调制和测量以具有某些品质的信号帧被认为是风噪声的结果。 在通信系统中进一步使用风噪声的帧被取消。
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公开(公告)号:US20110004470A1
公开(公告)日:2011-01-06
申请号:US12795188
申请日:2010-06-07
申请人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
发明人: Alon Konchitsky , Alberto D. Berstein , Sandeep Kulakcherla , William Martin Ribble , Kevin Fitzgerald , Don Seferovich
IPC分类号: G10L21/02
CPC分类号: G10L21/0208
摘要: A noisy signal is picked up by a microphone, digitized by an Analog to Digital Converter and fed to a processor for analysis and wind noise reduction. Most of noise reduction methods are based on the assumption that the interfering noise is stationary or slowly varying compared with speech. This assumption allows “learning” the characteristics of the noise between speech pauses and, based on a noise estimate, to build different filters that reduce the noise. In the case of wind noise this basic assumption is not valid. Wind noise is highly non-stationary, its power and spectral characteristics vary greatly. Because wind noise is not stationary, regular noise reduction methods cannot be used to reduce wind noise. For reducing wind noise effects in a device, the presence of wind should be detected reliably and then a novel approach presented here must be applied to eliminate the wind noise.
摘要翻译: 噪声信号由麦克风拾取,由模数转换器数字化,并馈送到处理器进行分析和减少风噪声。 大多数降噪方法是基于干扰噪声与语音相比是静止的或缓慢变化的假设。 该假设允许“学习”语音暂停之间的噪声的特性,并且基于噪声估计来建立减少噪声的不同滤波器。 在风噪声的情况下,这个基本假设是无效的。 风噪声非常不稳定,其功率和光谱特性差异很大。 由于风噪声不稳定,因此不能采用常规的降噪方法来减少风噪声。 为了减少设备中的风噪声影响,应该可靠地检测风的存在,然后必须采用这里提出的新颖方法来消除风噪声。
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公开(公告)号:US20110054889A1
公开(公告)日:2011-03-03
申请号:US12941827
申请日:2010-11-08
IPC分类号: G10L21/02
CPC分类号: G10L21/0208
摘要: The intelligibility of speech signals is improved in the many situations where a voice signal is communicated or stored. Means and methods are disclosed for developing a scheme with high voice signal intelligibility without sacrifice of voice quality. The disclosed method comprises certain steps, including, but not limited to: Learning the noise on near-end side and enhancing the far-end voice as a function of the noise level on the near-end side. The disclosed method and apparatus are especially useful to increase the intelligibility of the cell phone's loudspeaker output. The invention includes the processing of an input speech signal to generate an enhanced intelligent signal. In frequency domain, the FFT spectrum of the speech received from the far-end is modified in accordance with the LPC spectrum of the local background noise to generate an enhanced intelligent signal. In time domain, the speech is modified in accordance with the LPC coefficients of the noise to generate an enhanced intelligent signal.
摘要翻译: 在话音信号被传送或存储的许多情况下,语音信号的可懂度得到改善。 公开了用于开发具有高语音信号清晰度而不牺牲语音质量的方案的手段和方法。 所公开的方法包括某些步骤,包括但不限于:学习近端侧的噪声,并且增强远端声音作为近端侧的噪声电平的函数。 所公开的方法和装置对于提高手机的扬声器输出的可懂度是特别有用的。 本发明包括处理输入语音信号以产生增强的智能信号。 在频域中,根据本地背景噪声的LPC频谱修改从远端接收的语音的FFT频谱,以产生增强的智能信号。 在时域中,根据噪声的LPC系数修改语音以产生增强的智能信号。
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公开(公告)号:US20080312916A1
公开(公告)日:2008-12-18
申请号:US12139489
申请日:2008-06-15
申请人: Alon Konchitsky , Alberto D. Berstein , Hariharan Ganapathy Kathirvelu , Sandeep Kulakcherla , William Martin Ribble
发明人: Alon Konchitsky , Alberto D. Berstein , Hariharan Ganapathy Kathirvelu , Sandeep Kulakcherla , William Martin Ribble
IPC分类号: G10L21/00
CPC分类号: G10L21/0208
摘要: The intelligibility of speech signals is improved in the many situations where a voice signal is communicated or stored. Means and methods are disclosed for developing a scheme with high voice signal intelligibility without sacrifice of voice quality. The disclosed method comprises certain steps, including, but not limited to: Learning the noise on near-end side and enhancing the far-end voice as a function of the noise level on the near-end side. The disclosed method and apparatus are especially useful to increase the intelligibility of the cell phone's loudspeaker output. The invention includes the processing of an input speech signal to generate an enhanced intelligent signal. In frequency domain, the FFT spectrum of the speech received from the far-end is modified in accordance with the LPC spectrum of the local background noise to generate an enhanced intelligent signal. In time domain, the speech is modified in accordance with the LPC coefficients of the noise to generate an enhanced intelligent signal.
摘要翻译: 在话音信号被传送或存储的许多情况下,语音信号的可懂度得到改善。 公开了用于开发具有高语音信号清晰度而不牺牲语音质量的方案的手段和方法。 所公开的方法包括某些步骤,包括但不限于:学习近端侧的噪声,并且增强远端声音作为近端侧的噪声电平的函数。 所公开的方法和装置对于提高手机的扬声器输出的可懂度是特别有用的。 本发明包括处理输入语音信号以产生增强的智能信号。 在频域中,根据本地背景噪声的LPC频谱修改从远端接收的语音的FFT频谱,以产生增强的智能信号。 在时域中,根据噪声的LPC系数修改语音以产生增强的智能信号。
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公开(公告)号:US07817808B2
公开(公告)日:2010-10-19
申请号:US12176297
申请日:2008-07-18
申请人: Alon Konchitsky , Alberto D Berstein , Hariharan Ganapathy Kathirvelu , Sandeep Kulakcherla , William Martin Ribble
发明人: Alon Konchitsky , Alberto D Berstein , Hariharan Ganapathy Kathirvelu , Sandeep Kulakcherla , William Martin Ribble
IPC分类号: H04B15/00
CPC分类号: H04R1/406 , H04R3/005 , H04R2430/23 , H04R2499/11
摘要: A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.
摘要翻译: 具有高信噪比的清晰,高质量的语音信号通过使用紧密接近的两个麦克风的自适应降噪方案来实现。 该方法包括在高度定向模式下使用两个小型定向麦克风,然后应用自适应噪声消除算法来降低噪声。
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