公开(公告)号：US06804646B1

公开(公告)日：2004-10-12

申请号：US09646593

申请日：2000-09-19

Applicant: Tobias Schneider

Inventor： Tobias Schneider

IPC: G10L1520

CPC classification number: G10L21/0208 ,  G10L25/27

Abstract: A method and an apparatus for processing a sound signal in which a useful signal and an interference signal are specified, the sound signal being transformed into the frequency domain and a change in the profile of the frequency being represented by an envelope for at least one frequency over a time. By segmenting the envelope, a maximum is obtained for each segment, the smallest maximum, weighted by a factor, being subtracted from the sound signal. It is also possible to take account of the minimum for the purpose of reducing the interference signal.

Abstract translation: 一种用于处理其中指定有用信号和干扰信号的声音信号的方法和装置，所述声音信号被变换为频域，并且所述频率的轮廓的变化由用于至少一个频率的包络来表示 一段时间 通过对包络进行分割，对于每个片段获得最大值，从声音信号中减去最小最大值，由因子加权。 为了减少干扰信号，还可以考虑到最小值。

公开(公告)号：US06772118B2

公开(公告)日：2004-08-03

申请号：US10045683

申请日：2002-01-04

Applicant: Stephen C. Habermas ,  Ognjen Todic ,  Kai-Ten Feng ,  Jane F. MacFarlane

Inventor： Stephen C. Habermas ,  Ognjen Todic ,  Kai-Ten Feng ,  Jane F. MacFarlane

IPC: G10L1520

CPC classification number: G10L15/20

Abstract: An automated speech recognition filter is disclosed. The automated speech recognition filter device provides a speech signal to an automated speech platform that approximates an original speech signal as spoken into a transceiver by a user. In providing the speech signal, the automated speech recognition filter determines various models representative of a cumulative signal degradation of the original speech signal from various devices along a transmission signal path and a reception signal path between the transceiver and a device housing the filter. The automated speech platform can thereby provide an audio signal corresponding to a context of the original speech signal.

Abstract translation: 公开了一种自动语音识别滤波器。 自动语音识别滤波器装置向自动化语音平台提供语音信号，该语音平台近似由用户在收发器中说出的原始语音信号。 在提供语音信号时，自动语音识别滤波器确定表示沿着发送信号路径的各种装置的原始语音信号的累积信号劣化的各种模型以及收发机与容纳滤波器的装置之间的接收信号路径。 因此，自动语音平台可以提供对应于原始语音信号的上下文的音频信号。

公开(公告)号：US06725193B1

公开(公告)日：2004-04-20

申请号：US09660651

申请日：2000-09-13

Applicant: Thomas J. Makovicka

Inventor： Thomas J. Makovicka

IPC: G10L1520

CPC classification number: G10L21/02 ,  G10L2021/02082

Abstract: A voice recognition system for use with a communication system having an incoming line carrying an incoming signal from a first end to a second end operably attached to a speaker and the outgoing line carrying an outgoing signal from a microphone near the speaker. A first speech recognition unit (SRU) detects selected incoming words and a second SRU detect outgoing words. A comparator/signal generator compares the outgoing word with the incoming word and outputs the outgoing word when the outgoing word does not match the incoming word. The first SRU may be delayed relative to the second SRU. The SRU's may also search only for selected words in template, or may ignore words which are first detected by the other SRU. A signaler may also provide a signal indicating inclusion of one of the selected words in a known incoming signal with an SRU being responsive to that signal to ignore the included one command word in the template for a selected period of time.

Abstract translation: 一种与通信系统一起使用的语音识别系统，该通信系统具有输入线路，该输入线路将从第一端到第二端的输入信号可操作地连接到扬声器，并且输出线路承载来自扬声器附近的麦克风的输出信号。 第一语音识别单元（SRU）检测所选择的进入字，并且第二SRU检测输出字。 比较器/信号发生器将输出字与输入字进行比较，并在输出字与输入字不匹配时输出输出字。 第一SRU可以相对于第二SRU延迟。 SRU还可以仅搜索模板中的所选择的单词，或者可以忽略首先由另一个SRU检测到的单词。 信号器还可以提供指示在已知输入信号中包含所选择的单词之一的信号，其中SRU响应于该信号以忽略模板中包含的一个命令字在选定的时间段内。

公开(公告)号：US06671666B1

公开(公告)日：2003-12-30

申请号：US09381571

申请日：1999-08-24

Applicant: Keith M Ponting ,  Robert W Series ,  Michael J Tomlinson

Inventor： Keith M Ponting ,  Robert W Series ,  Michael J Tomlinson

IPC: G10L1520

CPC classification number: G10L15/065 ,  G10L15/142 ,  G10L15/20

Abstract: A recognition system (10) incorporates a filterbank analyser (16) producing successive data vectors of energy values for twenty-six frequency intervals in a speech signal. A unit (18) compensates for spectral distortion in each vector. Compensated vectors undergo a transformation into feature vectors with twelve dimensions and are matched with hidden Markov model states in a computer (24). Each matched model state has a mean value which is an estimate of the speech feature vector. A match inverter (28) produces an estimate of the speech data vector in frequency space by a pseudo-inverse transformation. It includes information which will be lost in a later transformation to frequency space. The estimated data vector is compared with its associated speech signal data vector, and infinite impulse response filters (44) average their difference with others. Averaged difference vectors so produced are used by the unit (18) in compensation of speech signal data vectors.

Abstract translation: 识别系统（10）包括滤波器组分析器（16），其在语音信号中产生二十六个频率间隔的能量值的连续数据向量。 单元（18）补偿每个矢量中的频谱失真。 补偿矢量经历了具有十二维度的特征向量的变换，并与计算机中的隐马尔可夫模型状态相匹配（24）。 每个匹配模型状态具有作为语音特征向量的估计的平均值。 匹配反相器（28）通过伪逆变换产生频率空间中的语音数据矢量的估计。 它包括将在以后的变换到频率空间中丢失的信息。 将估计的数据矢量与其相关联的语音信号数据矢量进行比较，无限脉冲响应滤波器（44）平均与其他数据矢量的差异。 如此产生的平均差分矢量由单元（18）用于补偿语音信号数据矢量。

公开(公告)号：US06473733B1

公开(公告)日：2002-10-29

申请号：US09452623

申请日：1999-12-01

Applicant: Dean McArthur ,  Jim Reilly

Inventor： Dean McArthur ,  Jim Reilly

IPC: G10L1520

CPC classification number: G10L21/0208 ,  G10L2021/02165 ,  G10L2021/02166

Abstract: An adaptive noise suppression system includes an input A/D converter, an analyzer, a filter, and a output D/A converter. The analyzer includes both feed-forward and feedback signal paths that allow it to compute a filtering coefficient, which is input to the filter. In these paths, feed-forward signal are processed by a signal to noise ratio estimator, a normalized coherence estimator, and a coherence mask. Also, feedback signals are processed by a auditory mask estimator. These two signal paths are coupled together via a noise suppression filter estimator. A method according to the present invention includes active signal processing to preserve speech-like signals and suppress incoherent noise signals. After a signal is processed in the feed-forward and feedback paths, the noise suppression filter estimator then outputs a filtering coefficient signal to the filter for filtering the noise out of the speech and noise digital signal.

Abstract translation: 自适应噪声抑制系统包括输入A / D转换器，分析器，滤波器和输出D / A转换器。 分析仪包括前馈和反馈信号路径，允许其计算滤波系数，滤波系数被输入到滤波器。 在这些路径中，前馈信号由信噪比估计器，标准化相干估计器和相干掩模进行处理。 此外，反馈信号由听觉掩模估计器处理。 这两个信号路径通过噪声抑制滤波器估计器耦合在一起。 根据本发明的方法包括有源信号处理以保持类似语音的信号并抑制非相干噪声信号。 在前馈和反馈路径中处理信号之后，噪声抑制滤波器估计器然后将滤波系数信号输出到滤波器，用于从语音和噪声数字信号中滤除噪声。

公开(公告)号：US06266636B1

公开(公告)日：2001-07-24

申请号：US09037998

申请日：1998-03-11

Applicant: Tetsuo Kosaka ,  Yasuhiro Komori

Inventor： Tetsuo Kosaka ,  Yasuhiro Komori

IPC: G10L1520

CPC classification number: G10L15/20 ,  G10L21/0216

Abstract: A process for removing additive noise due to the influence of ambient circumstances in a real-time manner in order to improve the precision of speech recognition which is performed in a real-time manner includes a converting process for converting a selected speech model distribution into a representative distribution, combining a noise model with the converted to generate speech model a noise superimposed speech model, performing a first likelihood calculation to recognize an input speech by using the noise superimposed speech model, converting the noise superimposed speech model to a noise adapted distribution that retains the relationship of the selected speech model, and performing a second likelihood calculation to recognize the input speech by using the noise adapted distribution.

Abstract translation: 为了提高以实时方式进行的语音识别的精度，为了提高实时地进行的语音识别的精度，由于周边环境的影响而实时地除去附加噪声的处理，包括将选择的语音模型分配变换为 将噪声模型与被转换为产生语音模型的噪声叠加语音模型相组合，执行第一似然计算以通过使用噪声叠加语音模型识别输入语音，将噪声叠加语音模型转换成噪声适应分布， 保留所选择的语音模型的关系，并且通过使用噪声适应分布来执行第二似然计算以识别输入语音。

公开(公告)号：US06778964B2

公开(公告)日：2004-08-17

申请号：US10217176

申请日：2002-08-12

Applicant: Dieter Geiger ,  Rainer Haberlander ,  Uwe Has

Inventor： Dieter Geiger ,  Rainer Haberlander ,  Uwe Has

IPC: G10L1520

CPC classification number: F24C15/2021 ,  G10L15/26 ,  G10L2015/223

Abstract: Improving voice recognition when there exist interference noises in a configuration with an electrically operated appliance, a voice input unit, and a voice processing unit that derives control signals for controlling functions of the appliance from spoken input instructions includes an operating status detection unit detecting the operating status of the household appliance or other noise sources and signals such detection results to the voice processing unit, the voice processing unit performing an interference noise correction only if a noise source is switched on.

Abstract translation: 当在电气设备，语音输入单元和语音处理单元的配置中存在干扰噪声时，改进语音识别，该控制信号从语音输入指令中导出用于控制设备功能的控制信号，包括操作状态检测单元，检测操作 家用电器或其他噪声源的状态，并将这种检测结果信号发送到语音处理单元，语音处理单元仅在噪声源被接通时执行干扰噪声校正。

公开(公告)号：US06615170B1

公开(公告)日：2003-09-02

申请号：US09519960

申请日：2000-03-07

Applicant: Fu-Hua Liu ,  Michael A. Picheny

Inventor： Fu-Hua Liu ,  Michael A. Picheny

IPC: G10L1520

CPC classification number: G10L25/78

Abstract: A system and method for voice activity detection, in accordance with the invention includes the steps of inputting data including frames of speech and noise, and deciding if the frames of the input data include speech or noise by employing a log-likelihood ratio test statistic and pitch. The frames of the input data are tagged based on the log-likelihood ratio test statistic and pitch characteristics of the input data as being most likely noise or most likely speech. The tags are counted in a plurality of frames to determine if the input data is speech or noise.

Abstract translation: 根据本发明的用于语音活动检测的系统和方法包括以下步骤：输入包括语音和噪声的数据，并且通过采用对数似然比检验统计量来确定输入数据的帧是否包括语音或噪声， 沥青。 输入数据的帧基于输入数据的对数似然比检验统计量和音调特性被标记为最可能是噪声或最可能的语音。 标签被计数在多个帧中以确定输入数据是语音还是噪声。

公开(公告)号：US06539350B1

公开(公告)日：2003-03-25

申请号：US09442392

申请日：1999-11-18

Applicant: Michael Walker

Inventor： Michael Walker

IPC: G10L1520

CPC classification number: G10L25/48 ,  G10L2025/783

Abstract: Speech level measurement is particularly significant for successful echo compensation in telecommunications systems, for noise suppression in a noisy environment, for example in military vehicles, or in speech recognition and in speech coding and decoding systems. A method is indicated which permits speech levels measurement only if features of speech are recognized and interferences and speech pauses are filtered out for the measurement. To this end, speech and pause detectors and a mean value generator are utilized, the time behavior of which is largely adapted to the perception capability of the human ear. Briefly spoken vowels thus are well detected, while nasal sounds or consonants are suppressed in the case of falling levels. A speech level measuring device is indicated which provides very accurate results in a short adaptation period.

Abstract translation: 语音电平测量对于电信系统中成功的回波补偿，例如在嘈杂环境中的噪声抑制，例如在军用车辆中，或在语音识别和语音编码和解码系统中尤其重要。 指示一种方法，其仅在语音特征被识别并且干扰和语音暂停被滤出用于测量时才允许语音水平测量。 为此，利用语音和暂停检测器和平均值发生器，其时间行为很大程度上适应于人耳的感知能力。 简单说出的元音因此被很好地发现，而鼻音或辅音在下降的情况下被压制。 指示语音电平测量装置，其在短的适应期内提供非常精确的结果。

公开(公告)号：US06487529B1

公开(公告)日：2002-11-26

申请号：US09426496

申请日：1999-10-26

Applicant: Gilles Miet

Inventor： Gilles Miet

IPC: G10L1520

CPC classification number: H04R3/00

Abstract: An audio processing device includes an analyzer and a filter. The analyzer extracts an envelope of a noise signal and derives therefrom noise envelope parameters. The filter has coefficients which vary in response to noise envelope parameters and filters a useful signal to form a filtered signal. The coefficients are varied so that the filter enhances frequency bands of the useful signal that correspond to frequency bands of the noise signal having a higher energy than a predetermined value.

Abstract translation: 音频处理装置包括分析器和滤波器。 分析仪提取噪声信号的包络，并从中得到噪声包络参数。 滤波器具有响应于噪声包络参数而变化的系数，并且滤波有用信号以形成滤波信号。 改变系数，使得滤波器增强对应于具有比预定值更高的能量的噪声信号的频带的有用信号的频带。