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1.发明授权Determining linear predictive coding filter parameters for encoding a voice signal 失效确定用于编码语音信号的线性预测编码滤波器参数公开(公告)号：US07013270B2
公开(公告)日：2006-03-14
申请号：US10924398
申请日：2004-08-23
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
IPC: G10L19/04
Abstract: The present invention is a method for determining linear predictive coding filter parameters for encoding a voice signal. The method includes sampling a voice signal, grouping the samples into a plurality of frames, generating a plurality of reflection coefficients for each frame of samples, quantizing the reflection coefficients, generating spectral coefficients from the quantized reflection coefficients, selecting a quantized reflection coefficient having the smallest log-spectral distance between a quantized spectrum, and an unquantized spectrum and, converting the selected quantized reflection coefficient to linear predictive coding (LPC) filter coefficient.
Abstract translation: 本发明是用于确定用于编码语音信号的线性预测编码滤波器参数的方法。 该方法包括采样语音信号，将样本分组成多个帧，为每个采样帧产生多个反射系数，量化反射系数，从量化反射系数产生频谱系数，选择具有 在量化频谱和未量化频谱之间的最小对数谱距离，并将选择的量化反射系数转换为线性预测编码（LPC）滤波器系数。
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2.发明授权Determining linear predictive coding filter parameters for encoding a voice signal 失效确定用于编码语音信号的线性预测编码滤波器参数公开(公告)号：US06782359B2
公开(公告)日：2004-08-24
申请号：US10446314
申请日：2003-05-28
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
IPC: G10L1904
Abstract: Linear predictive coding (LPC) filter parameters are determined for use in encoding a voice signal. Samples of a speech signal using a z-transform function are pre-emphasized. The pre-emphasized samples are analyzed to produce LPC reflection coefficients. The LPC reflection coefficients are quantized by a voiced quantizer and by an unvoiced quantizer producing sets of quantized reflection coefficients. Each set is converted into respective spectral coefficients. The set which produces a smaller lag-spectral distance is determined. The determined set is selected to encode the voice signal.
Abstract translation: 线性预测编码（LPC）滤波器参数被确定用于编码语音信号。 预先强调使用z变换函数的语音信号的样本。 分析预先强调的样本以产生LPC反射系数。 LPC反射系数由有声量化器和无声量化器量化，产生量化反射系数集。 每组被转换成各自的频谱系数。 确定产生较小滞后光谱距离的集合。 选择确定的集合来对语音信号进行编码。
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公开(公告)号：US06223152B1
公开(公告)日：2001-04-24
申请号：US09441743
申请日：1999-11-16
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
IPC: G01L1900
Abstract: To perform pitch analysis for encoding a speech signal, a speech signal is sampled. The sampled speech signal is spectrally whitened to produce a spectral residual signal. Samples of the spectral residual signal are collected and the collected samples are autocorrelated. Maximum values of the correlated result are determined. Gain values are determined based on at least in part the maximum values of the correlated result. The gain values are quantized using a codebook to produce a codebook index and an associated frame delay. The codebook index and the frame delay represent a pitch of the speech signal to facilitate encoding the speech signal.
Abstract translation: 为了执行用于编码语音信号的音调分析，对语音信号进行采样。 采样的语音信号被频谱白化以产生频谱残差信号。 收集光谱残差信号的样本，收集的样本是自相关的。 确定相关结果的最大值。 增益值至少部分地基于相关结果的最大值来确定。 增益值使用码本量化，以产生码本索引和相关帧延迟。 码本索引和帧延迟表示语音信号的音高以便于对语音信号进行编码。
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公开(公告)号：US5235670A
公开(公告)日：1993-08-10
申请号：US592330
申请日：1990-10-03
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
Abstract: The generation of multipulse excitation codes by digitizing an original speech, partitioning the digitized signal into a number of samples, pre-emphasizing the samples, producing linear predictive reflection coefficients from said samples, quantizing these reflection coefficients, converting the quantized reflection coefficients to spectral coefficients and subjecting the spectral coefficients to pitch analysis to obtain a spectral residual signal.
Abstract translation: 通过数字化原始语音来产生多脉冲激励码，将数字化信号划分成多个样本，预先强调样本，从所述样本产生线性预测反射系数，量化这些反射系数，将量化的反射系数转换成频谱系数 并对频谱系数进行音调分析以获得频谱残差信号。
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5.发明授权公开(公告)号：US07599832B2
公开(公告)日：2009-10-06
申请号：US11363807
申请日：2006-02-28
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
IPC: G10L19/12
Abstract: The present invention is a synthetic speech encoding device that produces a synthetic speech signal which closely matches an actual speech signal. The actual speech signal is digitized, and excitation pulses are selected by minimizing the error between the actual and synthetic speech signals. The preferred pattern of excitation pulses needed to produce the synthetic speech signal is obtained by using an excitation pattern containing a multiplicity of weighted pulses at timed positions. The selection of the location and amplitude of each excitation pulse is obtained by minimizing an error criterion between the synthetic speech signal and the actual speech signal. The error criterion function incorporates a perceptual weighting filter which shapes the error spectrum.
Abstract translation: 本发明是一种合成语音编码装置，其产生与实际语音信号紧密匹配的合成语音信号。 实际语音信号被数字化，并且通过最小化实际和合成语音信号之间的误差来选择激励脉冲。 通过使用在定时位置处包含多个加权脉冲的激励模式来获得产生合成语音信号所需的激励脉冲的优选模式。 通过最小化合成语音信号和实际语音信号之间的误差标准来获得每个激励脉冲的位置和振幅的选择。 错误标准函数包含形成误差谱的感知加权滤波器。
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Patent Agency Ranking
公开(公告)号：US06385577B1
公开(公告)日：2002-05-07
申请号：US09805634
申请日：2001-03-14
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
IPC: G10L1900
Abstract: A version of a speech signal and an output of a pitch synthesis filter and a linear predictive all-pole (LPC) filter is received. A system impulse response is produced based on in part the received pitch synthesis filter and LPC output. An excitation pulse location is determined so that the determined location minimizes an error between the speech signal version and the system impulse response. The speech signal is encoded with a representation of the determined location.
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公开(公告)号：US06006174A
公开(公告)日：1999-12-21
申请号：US950658
申请日：1997-10-15
Applicant: Daniel Lin , Brian M. McCarthy
Inventor： Daniel Lin , Brian M. McCarthy
Abstract: The generation of multipulse excitation codes by digitizing an original speech, partitioning the digitized signal into a number of samples, pre-emphasizing the samples, producing linear predictive reflection coefficients from said samples, quantizing these reflection coefficients, converting the quantized reflection coefficients to spectral coefficients and subjecting the spectral coefficients to pitch analysis to obtain a spectral residual signal.
Abstract translation: 通过数字化原始语音来产生多脉冲激励码，将数字化信号划分成多个样本，预先强调样本，从所述样本产生线性预测反射系数，量化这些反射系数，将量化的反射系数转换成频谱系数 并对频谱系数进行音调分析以获得频谱残差信号。
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公开(公告)号：US5630016A
公开(公告)日：1997-05-13
申请号：US614777
申请日：1996-03-07
Applicant: Kumar Swaminathan , Brian M. McCarthy
Inventor： Kumar Swaminathan , Brian M. McCarthy
CPC classification number: G10L19/012
Abstract: A digital discontinuous cellular communication system has a transmitter that transmits two frames of data following detection of voice inactivity. A receiver includes a comfort noise generator that uses the two frames of data to output noise to the speaker during period of voice inactivity. The comfort noise generator includes synthesis codebook with samples scaled by actual background noise and excitation codebook with samples filtered and scaled by the background noise that are combined to produce comfort noise having attributes and loudness level of the received background noise prior to interruption of transmission. The scaled signals are weighted to vary the loudness level and spectral attributes.
Abstract translation: 数字不连续的蜂窝通信系统具有在检测到语音不活动之后发送两帧数据的发射机。 接收机包括舒适噪声发生器，其在语音不活动期间使用两帧数据来向扬声器输出噪声。 舒适噪声发生器包括具有通过实际背景噪声缩放的样本的合成码本和激励码本，其中滤波和被背景噪声缩放的样本被组合以产生在传输中断之前具有接收到的背景噪声的属性和响度水平的舒适噪声。 加权比例的信号以改变响度级别和频谱属性。
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公开(公告)号：US5537509A
公开(公告)日：1996-07-16
申请号：US890747
申请日：1992-05-28
Applicant: Kumar Swaminathan , Brian M. McCarthy
Inventor： Kumar Swaminathan , Brian M. McCarthy
CPC classification number: H04W72/044 , G10L19/012 , H04B7/26 , H04B7/2643 , H04J3/172 , H04W28/06 , H04W72/0406
Abstract: A digital discontinuous cellular communication system has a transmitter that transmits two frames of data following detection of voice inactivity. A receiver includes a comfort noise generator that uses the two frames of data to output noise to the speaker during period of voice inactivity. The comfort noise generator includes synthesis codebook with samples scaled by actual background noise and excitation codebook with samples filtered and scaled by the background noise that are combined to produce comfort noise having attributes and loudness level of the received background noise prior to interruption of transmission. The scaled signals are weighted to vary the loudness level and spectral attributes.
Abstract translation: 数字不连续的蜂窝通信系统具有在检测到语音不活动之后发送两帧数据的发射机。 接收机包括舒适噪声发生器，其在语音不活动期间使用两帧数据来向扬声器输出噪声。 舒适噪声发生器包括具有通过实际背景噪声缩放的样本的合成码本和激励码本，其中滤波和被背景噪声缩放的样本被组合以产生在传输中断之前具有接收到的背景噪声的属性和响度水平的舒适噪声。 加权比例的信号以改变响度级别和频谱属性。
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公开(公告)号：US5333191A
公开(公告)日：1994-07-26
申请号：US51189
申请日：1993-04-22
Applicant: Brian M. McCarthy
Inventor： Brian M. McCarthy
Abstract: A method of operating a digital signal processor to detect DTMF tones in a digital voice telephone system in which the digitally encoded signals appearing on the telephone channel are decimated to compress the spectrum to be monitored for the appearance of call signalling tones. The signals received in a decimated block are "correlated" or convolved with one another on a forward and backward time-shifted basis and each forward and backward correlation product is summed to form the elements of a 5.times.5 modified covariance matrix. The modified covariance matrix exhibits the desirable property that its eigenvectors will be symmetric. Since all eigenvectors of the modified covariance matrix are orthogonal and the eigenvectors associated with the signal span the signal subspace, the signal subspace is orthogonal to the eigenvector associated with the noise. The dot product of the noise eigenvector with the signal subspace is set to zero. The roots of the resultant polynomial identify the frequencies of the DTMF tones, if in fact the same were present in the received signal. The noise and signal eigenvectors of the modified covariance matrix are more quickly and efficiently determined and advantageously, on a "real time" basis, by partitioning the modified covariance matrix into conjugate and anti-conjugate submatrices. The conjugate matrix is inverted and its eigenvalues determined, advantageously by applying the well-known power method. The largest eigenvalue of the inverted conjugate submatrix is related to the smallest eigenvalue of the original modified covariance matrix. When appropriate tones are being received this last-mentioned eigenvector should be the eigenvector associated with the noise. After determining the noise eigenvector the product of the signal space vector and the noise eigenvector is set to zero and the roots of the resultant polynomial are identified as the frequencies of the DTMF tones, advantageously through the use of a fast search technique.
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