公开(公告)号：US20100049342A1

公开(公告)日：2010-02-25

申请号：US12195837

申请日：2008-08-21

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

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

IPC分类号： G06F17/00

CPC分类号： G10L21/0388

摘要： A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope. This signal envelope can then be used to determine the at least one bounding frequency.

摘要翻译： 信号处理平台（300）呈现（101）要处理的信号，并识别（102）具有特定特征的信号部分（103），以自动确定可用于促进带宽扩展的至少一个边界频率 信号。 识别这些信号部分可以包括识别表现出至少预定水平能量的信号部分。 确定边界频率的步骤可以包括计算每个识别的信号部分的幅度谱，所述信号部分可被用于确定与通过频带内的对应的对应的测量平均度属于对应的归一化信号部分，从而提供对应的被检测的信号部分 。 确定边界频率可以包括累积这些被检测的信号部分的幅度谱，并且使用所得到的累积来估计相应的信号包络。 然后可以使用该信号包络来确定至少一个边界频率。

公开(公告)号：US5528723A

公开(公告)日：1996-06-18

申请号：US303271

申请日：1994-09-07

申请人： Ira A. Gerson ,  Mark A. Jasiuk

发明人： Ira A. Gerson ,  Mark A. Jasiuk

IPC分类号： G10L19/00 ,  G10L19/14 ,  G10L3/02

CPC分类号： G10L19/12 ,  G10L25/90

摘要： A digital speech coder utilizes harmonic noise weighting to overcome some limitations of low-rate CELP-type speech coders in reproducing voiced speech. In addition to a short term correction factor, which constitutes spectral noise weighting as known in the art, a long term pitch correction factor is utilized to provide harmonic noise weighting. The inclusion of harmonic noise weighting in a speech coder more efficiently utilizes noise-masking properties of a speech signal, allowing synthesis of a higher quality speech at a given bit rate.

摘要翻译： 数字语音编码器利用谐波噪声加权来克服在再现有声语音时低速率CELP型语音编码器的一些限制。 除了构成本领域已知的频谱噪声加权的短期校正因子之外，还使用长期的音调校正因子来提供谐波噪声加权。 在语音编码器中更高效地包含谐波噪声加权，利用语音信号的噪声掩蔽特性，允许以给定比特率合成较高质量的语音。

公开(公告)号：US20110112844A1

公开(公告)日：2011-05-12

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

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： 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：US20090201983A1

公开(公告)日：2009-08-13

申请号：US12027571

申请日：2008-02-07

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

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

IPC分类号： H04B1/66

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： 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：US5434947A

公开(公告)日：1995-07-18

申请号：US21364

申请日：1993-02-23

申请人： Ira A. Gerson ,  Mark A. Jasiuk ,  Matthew A. Hartman

发明人： Ira A. Gerson ,  Mark A. Jasiuk ,  Matthew A. Hartman

IPC分类号： G10L19/12 ,  H03H17/00 ,  H03H17/02 ,  H03M7/30 ,  H03M7/38 ,  H04B14/04 ,  G10L9/14

CPC分类号： G10L19/12

摘要： An Rth-order filter models the frequency response of multiple filters, to provide a filter which offers the control of multiple filters without the complexity of multiple filters. The Rth-order filter can be used as a spectral noise weighting filter or a combination of a short-term predictor filter and a spectral noise weighting filter, referred to as the spectrally noise weighted synthesis filter, depending on which embodiment is employed. In general, the method models the frequency response of L Pth-order filters by a single Rth-order filter, where the order R

摘要翻译： R阶滤波器对多个滤波器的频率响应进行建模，以提供一个滤波器，可提供多个滤波器的控制，而不需要多个滤波器的复杂性。 根据使用哪个实施例，R阶滤波器可以用作频谱噪声加权滤波器或短期预测器滤波器和被称为频谱噪声加权合成滤波器的频谱噪声加权滤波器的组合。 通常，该方法通过单个R阶滤波器对L P阶滤波器的频率响应建模，其中阶数R

公开(公告)号：US5241650A

公开(公告)日：1993-08-31

申请号：US870199

申请日：1992-04-13

申请人： Ira A. Gerson ,  Mark A. Jasiuk

发明人： Ira A. Gerson ,  Mark A. Jasiuk

IPC分类号： G10L19/14

CPC分类号： G10L19/26

摘要： An adaptive spectral postfilter in a synthesized speech platform has a denominator characteristic that corresponds to a preceding LPC filter stage, and a numerator characteristic that is developed as a function of the denominator characteristic through application of spectral smoothing techniques. This allows the numerator to track the denominator without the introduction of spectral distortion that would otherwise affect the processing in an adverse way.

摘要翻译： 合成语音平台中的自适应频谱后置滤波器具有对应于前面的LPC滤波器级的分母特征，以及通过应用频谱平滑技术作为分母特性的函数而发展的分子特征。 这允许分子跟踪分母而不引入光谱失真，否则会以不利的方式影响处理。

公开(公告)号：US08463412B2

公开(公告)日：2013-06-11

申请号：US12195837

申请日：2008-08-21

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

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

IPC分类号： G06F17/00

CPC分类号： G10L21/0388

摘要： A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope. This signal envelope can then be used to determine the at least one bounding frequency.

摘要翻译： 信号处理平台（300）呈现（101）要处理的信号，并识别（102）具有特定特征的信号部分（103），以自动确定可用于促进带宽扩展的至少一个边界频率 信号。 识别这些信号部分可以包括识别表现出至少预定水平能量的信号部分。 确定边界频率的步骤可以包括计算每个识别的信号部分的幅度谱，所述信号部分可被用于确定与通过频带内的对应的对应的测量平均度属于对应的归一化信号部分，从而提供对应的被检测的信号部分 。 确定边界频率可以包括累积这些被检测的信号部分的幅度谱，并且使用所得到的累积来估计相应的信号包络。 然后可以使用该信号包络来确定至少一个边界频率。

公开(公告)号：US08433582B2

公开(公告)日：2013-04-30

申请号：US12024620

申请日：2008-02-01

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

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

IPC分类号： G10L19/00

CPC分类号： G10L21/038 ,  G10L25/18 ,  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. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on a transition-band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

摘要翻译： 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。 基于在窄带带宽的预定的上限频率范围内的经处理的数字音频信号的过渡频带，估计与输入的数字音频信号对应的高频能量电平（103）。 基于高频带能级和对应于高频带能级的估计高频带，产生高频带数字音频信号（104）。

公开(公告)号：US20110112845A1

公开(公告)日：2011-05-12

申请号：US13008925

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

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).

公开(公告)号：US07792670B2

公开(公告)日：2010-09-07

申请号：US10964861

申请日：2004-10-14

申请人： Mark A. Jasiuk ,  Tenkasi V. Ramabadran ,  Udar Mittal ,  James P. Ashley ,  Michael J. McLaughlin

发明人： Mark A. Jasiuk ,  Tenkasi V. Ramabadran ,  Udar Mittal ,  James P. Ashley ,  Michael J. McLaughlin

IPC分类号： G10L19/00

CPC分类号： G10L19/09

摘要： A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.

摘要翻译： 本发明提供了语音编码系统中的预测方法和装置。 使用子样本分辨率延迟的1阶长期预测器（LTP）滤波器的方法被扩展到多抽头LTP滤波器，或者从另一个有利位置观察传统的整数样本分辨率多抽头 LTP过滤器扩展到使用子样本分辨率延迟。 多抽头LTP滤波器的这种新颖配方提供了优于现有技术的LTP滤波器配置的许多优点。 特别地，使用子样本分辨率定义滞后使得可以在内插滤波器使用的过采样因子的分辨率的限度内明确地建模具有分数分量的延迟值。 因此，这种多抽头LTP滤波器的系数在很大程度上不会对具有分数分量的延迟的影响进行建模。 因此，它们的主要功能是通过建模存在的周期程度和施加频谱整形来最大化LTP滤波器的预测增益。