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公开(公告)号：US5706392A
公开(公告)日：1998-01-06
申请号：US457517
申请日：1995-06-01
IPC分类号： G10L9/18
CPC分类号： G10L19/087
摘要： Simultaneous and temporal masking of digital speech data is applied to an MBE-based speech coding technique to achieve additional, substantial compression of coded speech over existing coding techniques, while enabling synthesis of coded speech with minimal perceptual degradation relative to the human auditory system. A real-time perceptual coder and decoder is disclosed in which speech may be sampled at 10 kHz, coded at an average rate of less than 2 bits/sample, and reproduced in a manner that is perceptually transparent to a human listener. The coder compresses speech segments that are inaudible due to simultaneous or temporal masking, while audible speech segments are not compressed.
摘要翻译： 数字语音数据的同时和时间屏蔽被应用于基于MBE的语音编码技术，以实现对现有编码技术的编码语音的额外的大量压缩，同时使得能够以相对于人类听觉系统的最小感知退化来合成编码语音。 公开了一种实时感知编码器和解码器，其中可以以10kHz采样语音，以小于2位/样本的平均速率进行编码，并以对人类听众感知地透明的方式再现。 编码器压缩由于同时或时间屏蔽而听不到的语音段，而声音段不被压缩。
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公开(公告)号：US4555598A
公开(公告)日：1985-11-26
申请号：US534204
申请日：1983-09-21
申请人： James L. Flanagan , Robert A. Kubli
发明人： James L. Flanagan , Robert A. Kubli
CPC分类号： H04R1/342
摘要： A directional acoustic transducer includes a plurality of acoustic paths each having first and second ends. The second end of each path terminates in the atmosphere. An electroacoustic device is attached to an acoustic cavity and the acoustic path first ends are coupled to the said acoustic cavity through an acoustic arrangement adapted to produce a predetermined transducer directional response pattern.
摘要翻译： 定向声换能器包括多个声道，每个声道具有第一和第二端。 每条路径的第二端在大气中终止。 电声装置附接到声腔，并且声路径第一端通过适于产生预定换能器定向响应图案的声学装置耦合到所述声腔。
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公开(公告)号：US4485484A
公开(公告)日：1984-11-27
申请号：US437290
申请日：1982-10-28
申请人： James L. Flanagan
发明人： James L. Flanagan
CPC分类号： H04R1/406 , H04R3/005 , H04R2201/401
摘要： A microphone arrangement focuses on a prescribed volume in a large room such as an auditorium. The arrangement includes a plurality of directable beam microphone structures. Each beam is directed to a prescribed location. The signals produced in the microphone structures are selectively adjusted to accept sounds from a predetermined volume surrounding the location and to reject sounds outside the prescribed volume.
摘要翻译： 麦克风安排的重点是在大房间（如礼堂）中的规定音量。 该装置包括多个可定向射束麦克风结构。 每个光束被引导到指定的位置。 选择性地调节麦克风结构中产生的信号以接收来自围绕该位置的预定音量的声音，并且拒绝规定音量外的声音。
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公开(公告)号：US4189627A
公开(公告)日：1980-02-19
申请号：US963926
申请日：1978-11-27
申请人： James L. Flanagan
发明人： James L. Flanagan
CPC分类号： H04R1/22
摘要： An electroacoustic transducer assembly adapted to filter sound waves in a digital communication system incorporates a plurality of tandemly arranged tubular members and a transducer. Each tubular member includes an apertured plate end, a tubular cavity and an open end. The open end of each tubular member is secured to the plate end of the adjacent tubular member to form a housing with a divided longitudinal passageway. The open end of the housing is secured to the transducer. Every tubular cavity is partitioned into longitudinal sections by structural elements to inhibit cross mode resonance. The apertures, cavity lengths and structural elements are dimensioned relative to the cavity cross sections to suppress passage of sound waves outside a predetermined frequency band.
摘要翻译： 适用于在数字通信系统中滤波声波的电声换能器组合包括多个串列布置的管状构件和换能器。 每个管状构件包括孔板端部，管状空腔和开口端。 每个管状构件的开口端固定到相邻管状构件的板端，以形成具有分开的纵向通道的壳体。 外壳的开口端固定在传感器上。 每个管状空腔通过结构元件分隔成纵向截面以抑制交叉模式共振。 孔，腔长度和结构元件相对于空腔横截面尺寸确定，以抑制声波在预定频带之外的通过。
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5.发明授权Method and apparatus for speaker recognition using selected spectral information 失效使用所选光谱信息进行扬声器识别的方法和装置
公开(公告)号：US5666466A
公开(公告)日：1997-09-09
申请号：US365598
申请日：1994-12-27
申请人： Qiguang Lin , James L. Flanagan , Ea-Ee Jan
发明人： Qiguang Lin , James L. Flanagan , Ea-Ee Jan
摘要： A method and apparatus are disclosed for robust, text-independent (and text-dependent) speaker recognition in which identification of a speaker is based on selected spectral information from the speaker's voice. Traditionally, speaker recognition systems (i) render a speech sample in the frequency domain to produce a spectrum, (ii) produce cepstrum coefficients from the spectrum, (iii) produce a codebook from the cepstrum coefficients, and (iv) use the codebook as the feature measure for comparing training speech samples with testing speech samples. The present invention, on the other hand, introduces the important and previously unknown step of truncating the spectrum prior to producing the cepstrum coefficients. Through the use of selected spectra as the feature measure for speaker recognition, the present invention has been shown to yield significant improvements in performance over prior art systems.
摘要翻译： 公开了用于稳健，文本无关（和文本相关）的扬声器识别的方法和装置，其中扬声器的识别基于来自扬声器的声音的所选频谱信息。 传统上，讲话者识别系统（i）在频域中呈现语音样本以产生频谱，（ii）从频谱产生倒谱系数，（iii）从倒谱系数产生码本，以及（iv）使用码本作为 将训练语音样本与测试语音样本进行比较的特征措施。 另一方面，本发明介绍了在产生倒谱系数之前截断频谱的重要且未知的步骤。 通过使用所选择的光谱作为说话者识别的特征量度，已经证明本发明在现有技术系统方面显着提高了性能。
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公开(公告)号：US4449190A
公开(公告)日：1984-05-15
申请号：US343238
申请日：1982-01-27
CPC分类号： G10L25/78 , H04B14/068
摘要： In an ADPCM system, improved detection of silence intervals in a speech signal is attained by detecting the level of the logarithm step-size signal (d.sub.n), which is representative of the energy of the speech samples. A speech pattern is converted into a sequence of adaptive digital codes. Intervals of silence in the pattern are detected and a digital code representative of each silence interval is generated. The adaptive digital codes and the silence interval codes are combined to form a digitally coded signal representative of the pattern. The conversion of the pattern to adaptive digital codes includes forming a signal corresponding to the adaptation step-size for each digital code. The silence interval detection includes producing first and second threshold signals. A silence interval signal is initiated when the adaptation step-size corresponding signal diminishes below the first threshold and the silence interval is terminated when the adaptation step-size corresponding signal increases above the second threshold after the silence interval initiation.
摘要翻译： 在ADPCM系统中，通过检测表示语音样本的能量的对数步长信号（dn）的电平来获得语音信号中的静音间隔的改进的检测。 语音模式被转换为自适应数字码的序列。 检测出图案中的静音间隔，并产生代表每个静默间隔的数字代码。 自适应数字码和静音间隔码被组合以形成表示图案的数字编码信号。 将图案转换为自适应数字码包括形成与每个数字码相应的适应步长相对应的信号。 静默间隔检测包括产生第一和第二阈值信号。 当自适应步长对应信号减小到低于第一阈值并且当自适应步长对应信号在静默间隔开始之后增加到高于第二阈值时，静音间隔终止，则开始静音间隔信号。
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7.发明授权
公开(公告)号：US4374304A
公开(公告)日：1983-02-15
申请号：US190993
申请日：1980-09-26
申请人： James L. Flanagan
发明人： James L. Flanagan
摘要： In a speech communication system, an input speech signal is partitioned into a plurality of subband portions. Responsive to each subband portion, a signal of lesser bandwidth representative of the subband portion is generated by dividing the instantaneous phase of the subband by an integer k. Where k=2, for example, the center frequency and bandwidth of each subband is halved. The lesser bandwidth subband portion representative signals are combined to form a compressed bandwidth signal representative of the input speech signal. A replica of the input speech signal is formed by partitioning the compressed bandwidth signal into subband portions thereof; converting each compressed signal subband portion into a signal representative of a subband of the input speech signal; and combining the converted subband representative signals into a single speech signal replica.
摘要翻译： 在语音通信系统中，将输入语音信号划分为多个子带部分。 响应于每个子带部分，通过将子带的瞬时相位除以整数k来生成表示子带部分的较小带宽的信号。 在k = 2的情况下，例如，每个子带的中心频率和带宽减半。 较小带宽子带部分代表性信号被组合以形成表示输入语音信号的压缩带宽信号。 输入语音信号的副本通过将压缩带宽信号划分成其子带部分而形成; 将每个压缩信号子带部分转换为表示输入语音信号的子带的信号; 并将转换的子带代表信号组合成单个语音信号副本。
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8.发明授权Method and apparatus including microphone arrays and neural networks for speech/speaker recognition systems 失效用于语音/扬声器识别系统的麦克风阵列和神经网络的方法和装置
公开(公告)号：US5737485A
公开(公告)日：1998-04-07
申请号：US399445
申请日：1995-03-07
申请人： James L. Flanagan , Qiguang Lin , Mazin Rahim , Chiwei Che
发明人： James L. Flanagan , Qiguang Lin , Mazin Rahim , Chiwei Che
摘要： A neural network is trained to transform distant-talking cepstrum coefficients, derived from a microphone array receiving speech from a speaker distant therefrom, into a form substantially similar to close-talking cepstrum coefficients that would be derived from a microphone close to the speaker, for providing robust hands-free speech and speaker recognition in adverse practical environments with existing speech and speaker recognition systems which have been trained on close-talking speech.
摘要翻译： 训练神经网络以将远程讲话倒频谱系数转换成与从扬声器靠近的麦克风相似的近似倒频谱系数，该距离系数从接收来自远离扬声器的扬声器的话筒的麦克风阵列转换成基本相似的形式，用于 在不利的实践环境中提供强大的免提语音和说话人识别，并且已经接受过讲话的语音训练的现有语音和扬声器识别系统。
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公开(公告)号：US4741038A
公开(公告)日：1988-04-26
申请号：US911989
申请日：1986-09-26
CPC分类号： G10K11/346 , H04R3/005 , H04R2201/401
摘要： A signal processing arrangement is connected to a microphone array to form at least one directable beam sound receiver. The directable beam sound receivers are adapted to receive sounds from predetermined locations in a prescribed environment such as auditorium. Signals representative of prescribed sound features received from the plurality of predetermined locations are generated and one or more of the locations is selected responsive to the sound feature signals. A plurality of directable beam sound receivers may be used to concurrently analyze sound features from the predetermined locations. Alternatively, one directable beam sound receiver may be used to scan the predetermined locations so that the sound feature signals therefrom are compared to sound features from a currently selected location.
摘要翻译： 信号处理装置连接到麦克风阵列以形成至少一个可定向波束声音接收器。 可定向射束声音接收器适于在诸如礼堂的规定环境中接收来自预定位置的声音。 产生代表从多个预定位置接收的规定声音特征的信号，并且响应于声音特征信号选择一个或多个位置。 可以使用多个可定向光束声音接收器来同时分析来自预定位置的声音特征。 或者，可以使用一个可定向光束声音接收器来扫描预定位置，使得其中的声音特征信号与来自当前选择的位置的声音特征进行比较。
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10.发明授权
公开(公告)号：US4653606A
公开(公告)日：1987-03-31
申请号：US714889
申请日：1985-03-22
申请人： James L. Flanagan
发明人： James L. Flanagan
摘要： An electroacoustic device comprises an array of electroacoustic transducer elements for producing a prescribed directional response pattern at a first frequency. Each element includes apparatus for restricting the frequency range of sound waves incident on said element so that the directional response pattern is invariant over a prescribed frequency band.
摘要翻译： 电声装置包括用于产生第一频率的规定方向响应图案的电声换能器阵列阵列。 每个元件包括用于限制入射在所述元件上的声波的频率范围的装置，使得定向响应图案在规定频带上不变。
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