公开(公告)号：US07047188B2

公开(公告)日：2006-05-16

申请号：US10290572

申请日：2002-11-08

申请人： Mark A. Jasiuk ,  James P. Ashley ,  Udar Mittal

发明人： Mark A. Jasiuk ,  James P. Ashley ,  Udar Mittal

IPC分类号： G10L19/04

CPC分类号： G10L19/083

摘要： A speech coder that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal. The speech coder generates a target vector based on an input signal. The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.

摘要翻译： 执行信号分析合成编码的语音编码器确定合成激励信号的多个构成分量的每个构成分量的增益参数。 语音编码器基于输入信号生成目标矢量。 语音编码器还产生与合成激励信号相关联的多个组成部分，其中多个组成部分的一个组成部分基于多个组成部分的另一组成部分的偏移版本。 语音编码器进一步基于目标矢量和多个构成分量来评估误差准则，以确定与多个构成分量的每个构成分量相关联的增益。

公开(公告)号：US5490230A

公开(公告)日：1996-02-06

申请号：US361474

申请日：1994-12-22

申请人： Ira A. Gerson ,  Mark A. Jasiuk

发明人： Ira A. Gerson ,  Mark A. Jasiuk

IPC分类号： G10L19/00 ,  G10L19/08 ,  G10L19/12 ,  G10L3/02

CPC分类号： G10L19/083 ,  G10L19/125

摘要： A speech coder and decoder methodology wherein pitch excitation and codebook excitation source energies are represented by parameters that are readily transmissible with minimal transmission capacity requirements. The parameters are the long term energy value, a short term correction factor which is applied to the long term energy value to match the short term energy, and proportionality factor(s) that specify the relative energy contribution of the excitation sources to the short term energy value.

摘要翻译： 语音编码器和解码器方法，其中音调激励和码本激励源能量由在最小传输容量要求下容易传播的参数表示。 参数是长期能量值，应用于长期能量值以匹配短期能量的短期校正因子，以及规定激励源与短期能量的相对能量贡献的比例因子 能量值。

公开(公告)号：US5359696A

公开(公告)日：1994-10-25

申请号：US214998

申请日：1994-03-21

申请人： Ira A. Gerson ,  Mark A. Jasiuk

发明人： Ira A. Gerson ,  Mark A. Jasiuk

IPC分类号： G10L19/00 ,  G10L19/12 ,  G10L9/18

CPC分类号： G10L19/12

摘要： A digital speech coder includes a long-term filter (124) having an improved sub-sample resolution long-term predictor (FIG. 5 ) which allows for subsample resolution for the lag parameter L. A frame of N samples of input speech vector s(n) is applied to an adder (510). The output of the adder (510) produces the output vector b(n) for the long term filter (124). The output vector b(n) is fed back to a delayed vector generator block (530) of the long-term predictor. The nominal long-term predictor lag parameter L is also input to the delayed vector generator block (530). The long-term predictor lag parameter L can take on non-integer values, which may be multiples of one half, one third, one fourth or any other rational fraction. The delayed vector generator (530) includes a memory which holds past samples of b(n). In addition, interpolated samples of b(n) are also calculated by the delayed vector generator (530) and stored in its memory, at least one interpolated sample being calculated and stored between each past sample of b(n). The delayed vector generator (530) provides output vector q(n) to the long-term multiplier block (520), which scales the long-term predictor response by the long-term predictor coefficient .beta.. The scaled output .beta.q(n) is then applied to the adder (510) to complete the feedback loop of the recursive filter (124).

摘要翻译： 数字语音编码器包括具有改进的子样本分辨率长期预测器（图5）的长期滤波器（124），其允许用于滞后参数L的子样本分辨率。输入语音向量s的N个样本的帧 （n）被施加到加法器（510）。 加法器（510）的输出产生用于长期滤波器（124）的输出向量b（n）。 输出向量b（n）被反馈给长期预测器的延迟向量生成器块（530）。 标称长期预测器滞后参数L也被输入到延迟向量发生器块（530）。 长期预测器滞后参数L可以采用非整数值，其可以是二分之一，三分之一，四分之一或任何其他有理分数的倍数。 延迟向量生成器（530）包括保存b（n）的过去样本的存储器。 另外，b（n）的内插样本也由延迟矢量发生器（530）计算并存储在其存储器中，至少一个内插样本被计算并存储在每个过去的样本b（n）之间。 延迟向量生成器（530）向长期乘法器块（520）提供输出向量q（n），长期乘数块（520）通过长期预测器系数β来缩放长期预测器响应。 然后将缩放的输出βq（n）加到加法器（510）以完成递归滤波器（124）的反馈回路。

公开(公告)号：US08527283B2

公开(公告)日：2013-09-03

申请号：US13008924

申请日：2011-01-19

申请人： Mark A. Jasiuk ,  Tenkasi V. Ramabadran

发明人： Mark A. Jasiuk ,  Tenkasi V. Ramabadran

IPC分类号： G10L19/00 ,  G10L19/02

CPC分类号： G10L21/038 ,  G10L25/21

摘要： A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. An estimate of the high-band energy level corresponding to the input digital audio signal is determined (103). Modification of the estimated high-band energy level is done based on an estimation accuracy and/or narrow-band signal characteristics (104). A high-band digital audio signal is generated based on the modified estimate of the high-band energy level and an estimated high-band spectrum corresponding to the modified estimate of the high-band energy level (105).

摘要翻译： 一种方法（100）包括接收（101）包括窄带信号的输入数字音频信号。 处理输入数字音频信号（102）以产生经处理的数字音频信号。 确定对应于输入数字音频信号的高频带能级的估计（103）。 基于估计精度和/或窄带信号特性来进行估计的高带能级的修改（104）。 基于高频带能级的修正估计和对应于高频带能级（105）的修改估计的估计高频带频谱，生成高频带数字音频信号。

公开(公告)号：US20090144062A1

公开(公告)日：2009-06-04

申请号：US11946978

申请日：2007-11-29

申请人： Tenkasi V. Ramabadran ,  Mark A. Jasiuk

发明人： Tenkasi V. Ramabadran ,  Mark A. Jasiuk

IPC分类号： G06F19/00

CPC分类号： G10L21/038

摘要： One provides (101) a digital audio signal having a corresponding signal bandwidth, and then provides (102) an energy value that corresponds to at least an estimate of out-of-signal bandwidth energy as corresponds to that digital audio signal. One then uses (103) the energy value to simultaneously determine both a spectral envelope shape and a corresponding suitable energy for the spectral envelope shape for out-of-signal bandwidth content as corresponds to the digital audio signal. By one approach, if desired, one then combines (104) (on, for example, a frame by frame basis) the digital audio signal with the out-of-signal bandwidth content to provide a bandwidth extended version of the digital audio signal to be audibly rendered to thereby improve corresponding audio quality of the digital audio signal as so rendered.

摘要翻译： 一个提供（101）具有相应信号带宽的数字音频信号，然后提供对应于该数字音频信号的至少对信号外带宽能量的估计的能量值（102）。 然后，对应于数字音频信号，然后使用（103）能量值同时确定频谱包络形状和对于用于超出信号带宽内容的频谱包络形状的相应合适的能量。 通过一种方法，如果需要，然后将数字音频信号与输出信号带宽内容（104）（例如，逐帧地组合）（104）以提供数字音频信号的带宽扩展版本， 可听见地渲染，从而提高如此呈现的数字音频信号的相应音频质量。

公开(公告)号：US07337110B2

公开(公告)日：2008-02-26

申请号：US10227725

申请日：2002-08-26

申请人： Mark A. Jasiuk

发明人： Mark A. Jasiuk

IPC分类号： G10L19/12

CPC分类号： G10L19/12 ,  G10L19/135 ,  G10L2019/0013

摘要： A codebook excited linear prediction coding system providing improved digital speech coding for high quality speech at low bit rates with side-by-side codebooks for segments of the modeled input signal to reduce the complexity of the codebook search. A linear predictive filter responsive to an input signal desired to be modeled is used for identifying a basis vector from a first codebook over predetermined intervals as a subset of the input signal. A long term predictor and a vector quantizer provide synthetic excitation of modeled waveform signal components corresponding to the input signal desired to be modeled from side-by-side codebooks by providing codevectors with concatenated signals identified from the basis vector over the predetermined intervals with respect to the side-by-side codebooks. Once a codevector is identified, the codebook at the next segment is searched and a concatenation of codevectors is provided by selecting up to but not including the current segment. The codevector is treated as an additional basis vector for the codebook search at the current segment. It is possible to significantly reduce the complexity of the VSELP codebook search by precomputing and storing the terms for the code search that do not change from segment to segment. Using these techniques, the complexity of searching a 45 bit VSELP codebook (N=40, M=45, M′=9, J=5) was found to be approximately equivalent to searching a traditionally structured 10 bit VSELP codebook (N=40, M=10, J=1). A concatenation of codevectors or carry-along basis vectors are formed as a concatenation of VSELP codevectors selected up to but not including the current segment.

摘要翻译： 码本激励线性预测编码系统，以低比特率的高质量语音提供改进的数字语音编码，并行编码用于建模输入信号的段，以降低码本搜索的复杂度。 响应于期望被建模的输入信号的线性预测滤波器被用于通过预定间隔从第一码本识别基本矢量作为输入信号的子集。 长期预测器和矢量量化器提供对与期望由并排码本建模的输入信号相对应的建模波形信号分量的综合激励，该码矢量相对于预定间隔，通过从基矢量识别的级联信号， 并排的码本。 一旦识别了码矢量，则搜索下一段的码本，并通过选择直到但不包括当前段来提供代码矢量的级联。 代码向量被视为当前段的码本搜索的附加基本向量。 通过预先计算和存储不从段到段的代码搜索的术语，可以显着降低VSELP码本搜索的复杂度。 使用这些技术，发现搜索45位VSELP码本（N = 40，M = 45，M'= 9，J = 5）的复杂度大致相当于搜索传统结构的10位VSELP码本（N = 40 ，M = 10，J = 1）。 代码矢量或携带方向矢量的级联形成为直到但不包括当前段的VSELP码矢量的级联。

公开(公告)号：US07254535B2

公开(公告)日：2007-08-07

申请号：US10882450

申请日：2004-06-30

申请人： William M. Kushner ,  Sara M. Harton ,  Mark A. Jasiuk

发明人： William M. Kushner ,  Sara M. Harton ,  Mark A. Jasiuk

IPC分类号： G10L21/00

CPC分类号： B63C11/26 ,  G10L15/20 ,  G10L21/0208 ,  G10L25/78 ,  G10L2021/03643

摘要： A method for equalizing a speech signal generated within a pressurized air delivery system, the method including the steps of: generating an inhalation noise model (1152) based on inhalation noise; receiving an input signal (802) that includes a speech signal; and equalizing the speech signal (1156) based on the noise model.

摘要翻译： 一种用于均衡加压空气输送系统内产生的语音信号的方法，所述方法包括以下步骤：基于吸入噪声产生吸入噪声模型（1152）; 接收包括语音信号的输入信号（802）; 以及基于噪声模型均衡语音信号（1156）。

公开(公告)号：US07170988B2

公开(公告)日：2007-01-30

申请号：US10694571

申请日：2003-10-27

申请人： Jonathan A. Gibbs ,  James P. Ashley ,  Halil Fikretler ,  Mark A. Jasiuk ,  Michael J. McLaughlin

发明人： Jonathan A. Gibbs ,  James P. Ashley ,  Halil Fikretler ,  Mark A. Jasiuk ,  Michael J. McLaughlin

IPC分类号： H04M7/00 ,  H04J3/16 ,  H04J3/22

CPC分类号： H04W88/181 ,  H04Q2213/13034 ,  H04Q2213/13166

摘要： A method of enhanced tandem communication is provided between at least a first portion of a network suitable for voice communications and a second portion of a network suitable for voice communications. During operation, two representations of an encoded signal are transmitted from the first portion of a network. The two representations comprise the encoded signal produced by a first codec and a parameter translation of the first encoded signal into an encoded signal compatible with a single common compressed voice codec (CCVC) format.

摘要翻译： 在适于语音通信的网络的至少第一部分和适于语音通信的网络的第二部分之间提供增强串联通信的方法。 在操作期间，从网络的第一部分发送编码信号的两个表示。 两个表示包括由第一编解码器产生的编码信号和第一编码信号的参数转换成与单个公共压缩语音编解码器（CCVC）格式兼容的编码信号。

公开(公告)号：US07139701B2

公开(公告)日：2006-11-21

申请号：US10882452

申请日：2004-06-30

申请人： Sara M. Harton ,  Mark A. Jasiuk ,  William M. Kushner

发明人： Sara M. Harton ,  Mark A. Jasiuk ,  William M. Kushner

IPC分类号： G10L19/00

CPC分类号： A62B18/08 ,  B63C11/26 ,  G10L2021/03643

摘要： A method for detecting and attenuating inhalation noise in a communication system coupled to a pressurized air delivery system, the method including the steps of: generating an inhalation noise model (912, 1012) based on inhalation noise; receiving an input signal (802) that includes inhalation noise; comparing (810) the input signal to the noise model to obtain a similarity measure; determining (854) a gain factor based on the similarity measure; and modifying (852) the input signal based on the gain factor, wherein the inhalation noise in the input signal is attenuated based on the gain factor.

摘要翻译： 一种用于在耦合到加压空气输送系统的通信系统中检测和减弱吸入噪声的方法，所述方法包括以下步骤：基于吸入噪声产生吸入噪声模型（912,1012）; 接收包括吸入噪声的输入信号（802）; 将输入信号与噪声模型进行比较（810）以获得相似性度量; 基于相似性度量确定（854）增益因子; 以及基于所述增益因子来修改（852）所述输入信号，其中所述输入信号中的吸入噪声基于所述增益因子衰减。

公开(公告)号：US07027980B2

公开(公告)日：2006-04-11

申请号：US10109151

申请日：2002-03-28

申请人： Tenkasi V. Ramabadran ,  Aaron M. Smith ,  Mark A. Jasiuk

发明人： Tenkasi V. Ramabadran ,  Aaron M. Smith ,  Mark A. Jasiuk

IPC分类号： G10L19/00 ,  G10L19/04

CPC分类号： G10L19/06 ,  G10L19/087

摘要： A system or method for modeling a signal, such as a speech signal, in which harmonic frequencies and amplitudes are identified and the harmonic magnitudes are interpolated to obtain spectral magnitudes at a set of fixed frequencies. An inverse transform is applied to the spectral magnitudes to obtain a pseudo auto-correlation sequence, from which linear prediction coefficients are calculated. From the linear prediction coefficients, model harmonic magnitudes are generated by sampling the spectral envelope defined by the linear prediction coefficients. A set of scale factors are then calculated as the ratio of the harmonic magnitudes to the model harmonic magnitudes and interpolated to obtain a second set of scale factors at the set of fixed frequencies. The spectral envelope magnitudes at the set of fixed frequencies are multiplied by the second set of scale factors to obtain new spectral magnitudes and the process is iterated to obtain final linear prediction coefficients. The signal is modeled by the linear prediction coefficients.