公开(公告)号：MX2010008288A

公开(公告)日：2010-08-31

申请号：MX2010008288

申请日：2009-02-05

Applicant: MOTOROLA INC

Inventor： JASIUK MARK A ,  RAMABADRAN TENKASI V

IPC: G10L21/02

Abstract: Un método (100) incluye recibir (101) una señal de audio digital de entrada que comprende una señal de banda angosta; la señal de audio digital de entrada es procesada (102) para generar una señal de audio digital procesada; se determina (103) un estimado del nivel de energía de banda alta correspondiente a una señal de audio digital de entrada de ancho de banda extendido; se realiza la modificación del nivel de energía de banda alta estimado con base en una precisión de la estimación y/o características de la señal de banda angosta (104); una señal de audio digital de banda alta es generada con base en el estimado modificado del nivel de energía de banda alta y un espectro de banda alta estimado correspondiente al estimado modificado del nivel de energía de banda alta (105).

公开(公告)号：DE60305907D1

公开(公告)日：2006-07-20

申请号：DE60305907

申请日：2003-02-14

Applicant: MOTOROLA INC

Inventor： RAMABADRAN TENKASI V ,  SMITH AARON M ,  JASIUK MARK A

IPC: G10L19/02 ,  G10L19/04 ,  G10L19/06 ,  G10L19/08

Abstract: 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.

公开(公告)号：BRPI0406937A

公开(公告)日：2006-01-03

申请号：BRPI0406937

申请日：2004-01-20

Applicant: MOTOROLA INC

Inventor： RAMABADRAN TENKASI V

IPC: G10L15/28 ,  G10L19/02 ,  G10L21/02 ,  G10L19/12

Abstract: A method and apparatus for noise suppression within a distributed speech recognition system is provided herein. Mel-frequency cepstral coefficients (MFCCs) values are converted to filter bank outputs (F'0 through F'22). The filter bank outputs are then used by a noise suppressor (303) for channel energy estimation, noise energy estimation, etc. Noise-suppression takes place on F'0 through F'22 and the noise-suppressed filter bank outputs F''0 through F''22 are converted back to MFCC values.

公开(公告)号：FR2739481B1

公开(公告)日：1999-02-26

申请号：FR9611654

申请日：1996-09-25

Applicant: MOTOROLA INC

Inventor： MCLAUGHLIN MICHAEL ,  RAMABADRAN TENKASI V

IPC: H04B14/04 ,  G10L9/00

Abstract: A signal that includes noise (301) is sampled to provide a plurality of digital information samples (303). A predetermined number of the digital information samples are grouped as a set (305). Noise suppression is performed on the signal using the following steps. One or more digital representations of silence is attached to the set, forming an extended set (401). A Fourier transform is performed on the extended set, yielding a set of frequency domain coefficients (403), at least some of which are scaled (405). An inverse Fourier transform is performed on the set of scaled frequency domain coefficients to provide a set of time domain samples (407), which are partially overlapped in time and added with a previously formed set of time domain samples (409 and 411), which result is provided with the non-overlapping time domain samples as a noise suppressed version of the signal (413).

公开(公告)号：EP1495465A4

公开(公告)日：2005-05-18

申请号：EP03745516

申请日：2003-02-14

Applicant: MOTOROLA INC

Inventor： RAMABADRAN TENKASI V ,  SMITH AARON M ,  JASIUK MARK A

IPC: G10L19/06 ,  G10L19/08 ,  G10L19/02 ,  G10L19/04

CPC classification number: G10L19/06 ,  G10L19/087

Abstract: A system or method for modeling a signal, such as a speech signal, wherein harmonic frequencies and amplitudes are identified (106) and the harmonic magnitudes are interpolated (110) to obtain spectral magnitudes at a set of fixed frequencies. An inverse transform is applied (112) to the spectral magnitudes to obtain a pseudo auto-correlation sequence, from which linear prediction coefficients are calculated (114). From the linear prediction coefficients, model harmonic magnitudes are generated by sampling the spectral envelope (118) defined by the linear prediction coefficients. A set of scale factors are then calculated (120) as the ratio of the harmonic magnitudes to the model harmonic magnitudes and interpolated to obtain a second set of scale factors (122) at the set of fixed frequencies. The spectral envelope magnitudes at the set of fixed frequencies (124) are multiplied by the second set of scale factors (126) to obtain new spectral magnitudes and the process is iterated to obtain final linear prediction coefficients.

公开(公告)号：WO2007062287A2

公开(公告)日：2007-05-31

申请号：PCT/US2006060656

申请日：2006-11-08

Applicant: MOTOROLA INC ,  CRUZ-ZENO EDGARDO M ,  PIKET JAMES B ,  RAMABADRAN TENKASI V

Inventor： CRUZ-ZENO EDGARDO M ,  PIKET JAMES B ,  RAMABADRAN TENKASI V

IPC: H04M9/08 ,  A61F11/06 ,  G10K11/16 ,  H03B29/00

CPC classification number: H04M9/082

Abstract: An echo canceling circuit comprising a double talk detector, an upper band signal filter configured to pass only near-end upper band signals to the double talk detector and remove lower band signals, an adaptive filter circuit, a control circuit operatively coupled to the double talk detector and to the adaptive filter circuit, and a threshold estimator configured to iteratively calculate an upper adaptive decision threshold value and a lower adaptive decision threshold value. The double talk detector declares near-end speech to be present if an estimated power level of the upper band signals exceeds the upper adaptive decision threshold value, and declares the near-end speech to be absent if the estimated power level of the upper band signals falls below the lower adaptive decision threshold value for a predetermined number of iterative cycles.

Abstract translation: 一种回声消除电路，包括双通话检测器，上频带信号滤波器，其被配置为仅将近端高频带信号传递到双方通话检测器并去除低频带信号;自适应滤波器电路;可操作地耦合到双重通话的控制电路 检测器和自适应滤波器电路;以及阈值估计器，被配置为迭代地计算上自适应判决阈值和较低自适应判决阈值。 如果高频信号的估计功率电平超过上限自适应判定阈值，双通话检测器声明近端语音存在，并且如果高频信号的估计功率电平则声明近端语音不存在 在预定数量的迭代循环下降到低于自适应判决阈值以下。

公开(公告)号：WO2007067837A3

公开(公告)日：2008-06-05

申请号：PCT/US2006060935

申请日：2006-11-15

Applicant: MOTOROLA INC ,  MA CHANGXUE C ,  CHENG YAN M ,  NOWLAN STEVEN J ,  RAMABADRAN TENKASI V

Inventor： MA CHANGXUE C ,  CHENG YAN M ,  NOWLAN STEVEN J ,  RAMABADRAN TENKASI V

IPC: G10L15/04

CPC classification number: G10L25/69 ,  G10L15/26

Abstract: A method and apparatus are provided for reproducing a speech sequence of a user through a communication device of the user. The method includes the steps of detecting a speech sequence from the user through the communication device, recognizing a phoneme sequence within the detected speech sequence and forming a confidence level of each phoneme within the recognized phoneme sequence. The method further includes the steps of audibly reproducing the recognized phoneme sequence for the user through the communication device and gradually highlighting or degrading a voice quality of at least some phonemes of the recognized phoneme sequence based upon the formed confidence level of the at least some phonemes.

Abstract translation: 提供了一种用于通过用户的通信设备再现用户的语音序列的方法和装置。 该方法包括以下步骤：通过通信设备检测来自用户的语音序列，识别检测到的语音序列内的音素序列，并形成识别的音素序列内每个音素的置信度。 该方法还包括以下步骤：通过通信设备可听地再现用户的识别音素序列，并且基于形成的至少一些音素的置信水平逐渐突出或降低所识别的音素序列的至少一些音素的语音质量 。

公开(公告)号：WO2004095420A3

公开(公告)日：2005-06-09

申请号：PCT/US2004008646

申请日：2004-03-19

Applicant: MOTOROLA INC ,  RAMABADRAN TENKASI V ,  SORIN ALEXANDER

Inventor： RAMABADRAN TENKASI V ,  SORIN ALEXANDER

IPC: G10L20060101 ,  G10L11/04

CPC classification number: G10L25/90

Abstract: A system, computer readable medium, and method for sampling a speech signal; dividing the sampled speech signal into overlapped frames; extracting first pitch information from a frame using frequency domain analysis; providing at least one pitch candidate, each being associated with a spectral score, from the first pitch information, each of the at least one pitch candidate representing a possible pitch estimate for the frame; extracting second pitch information from the frame using a time domain analysis; providing a correlation score for the at least one pitch candidate from the second pitch information; and selecting one of the at least one pitch candidate to represent the pitch estimate of the frame. The system, computer readable medium, and method are suitable for speech coding and for distributed speech recognition.

Abstract translation: 一种用于对语音信号进行采样的系统，计算机可读介质和方法; 将采样语音信号划分成重叠帧; 使用频域分析从帧中提取第一音调信息; 从所述第一音调信息提供与频谱分数相关联的至少一个音调候选者，所述至少一个音调候选中的每一个表示所述帧的可能音调估计; 使用时域分析从帧中提取第二音调信息; 从所述第二音调信息提供所述至少一个音调候选的相关得分; 以及选择所述至少一个音调候选中的一个以表示所述帧的音调估计。 该系统，计算机可读介质和方法适用于语音编码和分布式语音识别。

公开(公告)号：WO2007047037A3

公开(公告)日：2009-04-09

申请号：PCT/US2006037408

申请日：2006-09-26

Applicant: MOTOROLA INC ,  JASIUK MARK A ,  RAMABADRAN TENKASI V

Inventor： JASIUK MARK A ,  RAMABADRAN TENKASI V

IPC: G10L19/04 ,  G10L19/08 ,  G10L19/12

CPC classification number: G10L19/26

Abstract: A speech communication system provides a speech encoder [100] that generates a set of coded parameters representative of the desired speech signal characteristics. The speech communication system also provides a speech decoder [200] that receives the set of coded parameters to generate reconstructed speech. The speech decoder includes an equalizer [204] that computes a matching set of parameters from the reconstructed speech [301] generated by the speech decoder [200], undoes the set of characteristics corresponding to the computed set of parameters, and imposes the set of characteristics corresponding to the coded set of parameters, thereby producing equalized reconstructed speech [306].

Abstract translation: 语音通信系统提供语音编码器[100]，其产生表示所需语音信号特性的编码参数集合。 语音通信系统还提供一种语音解码器[200]，其接收编码参数集合以产生重构语音。 语音解码器包括：均衡器[204]，其从语音解码器[200]生成的重构语音[301]中计算一组匹配的参数，撤销对应于所计算的参数集合的一组特征，并且将 特征对应于编码的参数集合，从而产生均衡的重构语音[306]。

公开(公告)号：WO2007062287A8

公开(公告)日：2008-08-21

申请号：PCT/US2006060656

申请日：2006-11-08

Applicant: MOTOROLA INC ,  CRUZ-ZENO EDGARDO M ,  PIKET JAMES B ,  RAMABADRAN TENKASI V

Inventor： CRUZ-ZENO EDGARDO M ,  PIKET JAMES B ,  RAMABADRAN TENKASI V

IPC: H04M9/08

CPC classification number: H04M9/082

Abstract: An echo canceling circuit comprising a double talk detector, an upper band signal filter configured to pass only near-end upper band signals to the double talk detector and remove lower band signals, an adaptive filter circuit, a control circuit operatively coupled to the double talk detector and to the adaptive filter circuit, and a threshold estimator configured to iteratively calculate an upper adaptive decision threshold value and a lower adaptive decision threshold value. The double talk detector declares near-end speech to be present if an estimated power level of the upper band signals exceeds the upper adaptive decision threshold value, and declares the near-end speech to be absent if the estimated power level of the upper band signals falls below the lower adaptive decision threshold value for a predetermined number of iterative cycles.

Abstract translation: 一种回声消除电路，包括双方讲话检测器，被配置为仅将近端高频带信号传递到双方讲话检测器并移除较低频带信号的高频带信号滤波器，自适应滤波器电路，操作性地耦合到双方讲话的控制电路 检测器和自适应滤波器电路，以及阈值估计器，被配置为迭代地计算上自适应判决阈值和下自适应判决阈值。 如果所述较高频带信号的估计功率电平超过所述较高自适应判决阈值，则所述双重通话检测器宣告近端语音存在，并且如果所述较高频带信号的所估计功率电平宣告所述近端语音不存在， 低于较低的自适应判定阈值达预定数量的迭代周期。