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公开(公告)号：US4737976A
公开(公告)日：1988-04-12
申请号：US59978
申请日：1985-09-03
申请人： David E. Borth , Ira A. Gerson , Richard J. Vilmur
发明人： David E. Borth , Ira A. Gerson , Richard J. Vilmur
IPC分类号： G10L15/00 , G10L15/02 , G10L15/20 , G10L21/02 , H04M1/27 , H04M1/274 , H04M1/2745 , H04M1/60 , H04B1/40
CPC分类号： H04M1/271 , G10L15/00 , H04M1/6041 , H04R2499/13
摘要： An improved hands-free user-interactive control and dialing system is disclosed for use with a speech communications device. The control system (400) includes a dynamic noise suppressor (410), a speech recognizer (420) for implementing voice-control, a device controller (430) responsive to the speech recognizer for controlling operating parameters of the speech communications device (450) and for producing status information representing the operating status of the device, and a speech synthesizer (440) for providing reply information to the user as to the speech communications device operating status. In a mobile radiotelephone application, the spectral subtraction noise suppressor (414) is configured to improve the performance of the speech recognizer (424), the voice quality of the transmitted audio (417), and the audio switching operation of the vehicular speakerphone (460). The combination of noise processing, speech recognition, and speech synthesis provides a substantial improvement to prior art control systems.
摘要翻译： PCT No.PCT / US85 / 01672 Sec。 371日期：1985年9月3日 102（e）1985年9月3日PCT PCT公布1985年9月3日PCT公布。 公开号WO87 / 01546 日期为1987年3月12日。公开了一种用于语音通信设备的改进的免提用户交互式控制和拨号系统。 控制系统（400）包括用于实现语音控制的动态噪声抑制器（410），语音识别器（420），响应于语音识别器控制语音通信设备（450）的操作参数的设备控制器（430） 以及用于产生表示设备的操作状态的状态信息，以及用于向用户提供关于语音通信设备操作状态的回复信息的语音合成器（440）。 在移动无线电话应用中，频谱减法噪声抑制器（414）被配置为提高语音识别器（424）的性能，传输音频的语音质量（417）和车载扬声器（460）的音频切换操作 ）。 噪声处理，语音识别和语音合成的组合为现有技术的控制系统提供了实质性的改进。
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2.发明授权Method and apparatus for synthesizing speech without voicing or pitch information 失效用于合成语音而不具有声音或倾斜信息的方法和装置
公开(公告)号：US5133010A
公开(公告)日：1992-07-21
申请号：US484008
申请日：1990-02-21
CPC分类号： G10L19/02
摘要： A channel bank speech synthesizer for reconstructing speech from externally-generated acoustic feature information without using externally-generated voicing or pitch information is disclosed. An N-channel pitch-excited channel bank synthesizer (340) is provided having a first low-frequency group of channel gain values (1 to M) and a second high-frequency group of channel gain values (+1 to N). The first group controls a first group of amplitude modulators (950) excited by a periodic pitch pulse source (920), and the second group controls amplitude modulators excited by a noise source (930). Both groups of modulated excitation signals are applied to the bandpass filters (960) to reconstruct the speech channels, and then combined at the summation network (970) to form a reconstructed synthesized speech signal. Additionally, the pitch pulse source (920) varies the pitch pulse period such that the pitch pulse rate decreases over the length of the word.
摘要翻译： 公开了一种用于从外部产生的声音特征信息重建语音而不使用外部产生的发音或音调信息的频道组语音合成器。 提供了具有第一低频组的信道增益值（1至M）和第二高频组的信道增益值（+1至N）的N沟道音调激励信道组合成器（340）。 第一组控制由周期性音调脉冲源（920）激发的第一组幅度调制器（950），并且第二组控制由噪声源（930）激发的幅度调制器。 两组调制激励信号被施加到带通滤波器（960）以重建语音信道，然后在求和网络（970）处组合以形成重构的合成语音信号。 此外，音调脉冲源（920）改变音调脉冲周期，使得音调脉冲频率在单词的长度上减小。
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公开(公告)号：US4630305A
公开(公告)日：1986-12-16
申请号：US750941
申请日：1985-07-01
CPC分类号： G10L21/0208 , G10L2021/02168 , G10L25/27 , H04R2225/43 , H04R25/505
摘要： An automatic gain selector is disclosed for use with a noise suppression system which performs speech quality enhancement upon a noisy speech signal available at the input to generate a noise-suppressed speech signal at the output by spectral gain modification. The channel gain controller (240) of the present invention produces a modification signal (245), comprised of individual channel gain values, for application to a channel gain modifier (250). A particular gain table set is automatically selected from one of a plurality of gain tables (450) by a selector switch (470) and a noise level quantizer (440) in response to a multi-channel noise parameter, such as the overall average background noise level of the input signal. Then the individual channel gain values (455) are obtained from the particular gain table set in response to the individual channel signal-to-noise ratio estimate (235). Hence, each individual channel gain value is selected as a function of (a) the channel number, (b) the current channel SNR estimate, and (c) the overall average background noise level. The automatic gain selector further includes a gain smoothing filter (460) for smoothing these noise suppression gain factors on a per-sample basis thereby improving noise flutter performance caused by step discontinuities in frame-to-frame gain changes.
摘要翻译： 公开了一种与噪声抑制系统一起使用的自动增益选择器，该噪声抑制系统在输入端可用的噪声语音信号上执行语音质量增强，以通过频谱增益修改在输出端产生噪声抑制语音信号。 本发明的信道增益控制器（240）产生一个包括各个信道增益值的修改信号（245），用于应用于信道增益修改器（250）。 响应于诸如总体平均背景的多通道噪声参数，选择器开关（470）和噪声电平量化器（440）从多个增益表（450）中的一个自动选择特定增益表集合 输入信号的噪声电平。 然后，从响应于各个信道信噪比估计（235）设置的特定增益表获得各个信道增益值（455）。 因此，根据（a）信道号，（b）当前信道SNR估计和（c）整体平均背景噪声电平来选择每个单独信道增益值。 自动增益选择器还包括增益平滑滤波器（460），用于在每采样的基础上平滑这些噪声抑制增益因子，从而改善由帧间增益变化中的步长不连续性引起的噪声颤动性能。
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公开(公告)号：US4630304A
公开(公告)日：1986-12-16
申请号：US750572
申请日：1985-07-01
申请人： David E. Borth , Ira A. Gerson , Richard J. Vilmur
发明人： David E. Borth , Ira A. Gerson , Richard J. Vilmur
IPC分类号： G10L11/00 , G10K11/178 , G10L19/02 , G10L21/02 , H03G3/32 , H04R25/00 , H04R27/00 , H04B15/00
CPC分类号： G10L21/0208 , G10K11/1782 , G10K2210/108 , G10K2210/3011 , G10K2210/3012 , G10K2210/3023 , H04R2225/43 , H04R25/505
摘要： An improved background noise estimator (320) is disclosed for use with a noise suppression system (300) for generating an estimate of the background noise power spectral density provided to noise suppressor (310), which performs speech quality enhancement upon the pre-processed speech-plus-noise signal available at the input to generate a clean post-processed speech signal at the output. Background noise estimator (320) utilizes an energy valley detector based upon post-processed speech to perform the speech/noise classification, and a noise spectral estimator based upon pre-processed speech to generate an estimate of the background noise power spectral density. As a result, the background noise estimate supplied to the noise suppressor is a more accurate measurement of the background noise energy, since it is performed during a more accurate determination of the occurrences of pauses in the speech.
摘要翻译： 公开了改进的背景噪声估计器（320），用于与噪声抑制系统（300）一起使用，用于产生提供给噪声抑制器（310）的背景噪声功率谱密度的估计，其对预处理语音执行语音质量增强 在输入端可用的-plus-noise信号在输出端产生干净的后处理语音信号。 背景噪声估计器（320）利用基于后处理语音的能量谷检测器来执行语音/噪声分类，以及基于预处理语音的噪声谱估计器，以产生背景噪声功率谱密度的估计。 结果，提供给噪声抑制器的背景噪声估计是背景噪声能量的更精确的测量，因为它在更准确地确定语音中的暂停的发生时执行。
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公开(公告)号：US4628529A
公开(公告)日：1986-12-09
申请号：US750942
申请日：1985-07-01
申请人： David E. Borth , Ira A. Gerson , Richard J. Vilmur
发明人： David E. Borth , Ira A. Gerson , Richard J. Vilmur
CPC分类号： G10L21/0208 , H04R2225/43 , H04R25/505
摘要： An improved noise suppression system (400) is disclosed which performs speech quality enhancement upon speech-plus-noise signal available at the input (205) to generate a clean speech signal at the output (265) by spectral gain modification. The noise suppression system of the present invention includes a background noise estimator (420) which generates and stores an estimate of the background noise power spectral density based upon pre-processed speech (215), as determined by the detected minima of the post-processed speech energy level. This post-processed speech (255) may be obtained directly from the output of the noise suppression system, or may be simulated by multiplying the pre-processed speech energy (225) by the channel gain values of the modification signal (245). This technique of implementing post-processed signal to generate the background noise estimate (325) provides a more accurate measurement of the background noise energy since it is based upon much cleaner speech signal. As a result, the present invention performs acoustic noise suppression in high ambient noise backgrounds with significantly less voice quality degradation.
摘要翻译： 公开了一种改进的噪声抑制系统（400），其通过在输入端（205）处可用的语音加噪声信号来执行语音质量增强，以通过频谱增益修改在输出端（265）产生干净的语音信号。 本发明的噪声抑制系统包括背景噪声估计器（420），该背景噪声估计器（420）根据检测到的后处理的最小值来确定基于预处理语音（215）的背景噪声功率谱密度的估计 言语能量水平。 该后处理语音（255）可以直接从噪声抑制系统的输出获得，或者可以通过将预处理的语音能量（225）乘以修改信号（245）的信道增益值来模拟。 实现后处理信号以产生背景噪声估计的这种技术（325）提供背景噪声能量的更精确的测量，因为它基于更清洁的语音信号。 结果，本发明在高环境噪声背景下执行噪声抑制，语音质量降低明显减少。
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公开(公告)号：US4811404A
公开(公告)日：1989-03-07
申请号：US103857
申请日：1987-10-01
CPC分类号： G10L21/0208 , G10L2021/02085 , G10L2021/02168 , G10L2025/786 , G10L2025/937
摘要： An improved noise suppression system (800) is disclosed which performs speech quality enhancement upon the speech-plus-noise signal available at the input (205) to generate a clean speech signal at the output (265) by spectral gain modification. The improvements of the present invention include the addition of a signal-to-noise ratio (SNR) threshold mechanism (830) to reduce background noise flutter by offsetting the gain rise of the gain tables until a certain SNR threshold is reached, the use of a voice metric calculator (810) to produce more accurate background noise estimates via performing the update decision based on the overall voice-like characteristics in the channels and the time interval since the last update, and the use of a channel SNR modifier (820) to provide immunity to narrowband noise bursts through modification of the SNR estimates based on the voice metric calculation and the channel energies.
摘要翻译： 公开了一种改进的噪声抑制系统（800），其在输入（205）上可用的语音加噪声信号上执行语音质量增强，以通过频谱增益修改在输出端（265）产生干净的语音信号。 本发明的改进包括增加信噪比（SNR）阈值机制（830）以通过抵消增益表的增益上升来减小背景噪声颤动，直到达到某个SNR阈值，使用 语音度量计算器（810），用于通过基于上一次更新以来的信道和时间间隔中的总体语音特征执行更新判定，以及使用信道SNR修改器（820）来产生更准确的背景噪声估计， 通过基于语音度量计算和信道能量修改SNR估计来提供对窄带噪声突发的抗扰性。
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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
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。
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公开(公告)号：US4905288A
公开(公告)日：1990-02-27
申请号：US262173
申请日：1988-10-18
申请人： Ira A. Gerson , Brett L. Lindsley
发明人： Ira A. Gerson , Brett L. Lindsley
CPC分类号： G10L15/00
摘要： The present invention describes a method and arrangement for reducing a sequence of initial frames into a reduced set of representative frames by combining the initial frames into a plurality of representative frames, the combining process including generating a distortion measure associated with each representative frame and comparing each distortion measure to a distortion threshold. From these representative frames, a set of mutually exclusive frames is determined to minimize the number of representative frames, whereby each representative frame in the set represents a unique set of contiguous initial frames and has an associated distortion measure which does not exceed the distortion threshold.
摘要翻译： 本发明描述了一种用于通过将初始帧组合成多个代表性帧来将初始帧序列减少为缩小集合的代码帧的方法和装置，所述组合处理包括生成与每个代表性帧相关联的失真度量， 失真测量到失真阈值。 从这些代表性的帧中，确定一组相互排列的帧以最小化代表帧的数量，由此集合中的每个代表帧表示连续的初始帧的唯一集合，并且具有不超过失真阈值的相关联的失真度量。
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公开(公告)号：US4354248A
公开(公告)日：1982-10-12
申请号：US98093
申请日：1979-11-28
CPC分类号： H04Q1/4575 , H04L27/30
摘要： A multifrequency tone receiver is disclosed for detecting simultaneous tone signals in a sampled digital signal. The tone receiver includes a microprogrammed sequence controller, a time-multiplexed digital filter and a signal processing microcomputer. For each sample of the digital signal, the sequence controller is programmed to time multiplex the digital filter for performing three cascaded second order filtering operations (two bandpass filter operations and one low pass filter operation) for each of six tone signals to provide corresponding energy estimates and one additioal filtering operation to provide a total energy estimate. The signal processing microcomputer processes a number of sets of the seven energy estimates and provides an indication when a multifrequency toner pair has been detected. The digital filter, when enabled by a filter start signal from the sequence controller, asynchronously performs a single multiplication-like filtering operation to implement each second-order filter, and provides a filter done signal upon completion of the filtering operation. Full-wave rectifying capability is provided during low pass filtering operations by logically complementing the digital filter input signal. Limit cycles may be suppressed in the digital filter output signal by rounding the output signal and clamping positive and negative overflows to the largest allowable positive and negative signals, respectively. The tone receiver may be advantageously utilized in a PCM communication system for detecting multifrequency tone signalling used for dialing and supervisory control. Moreover, the inventive tone reciver may be adapted to receive many different types of tone signalling simply by changing firmware therewithin.
摘要翻译： 公开了一种用于检测采样数字信号中的同时音调信号的多频音频接收机。 音调接收机包括微程序序列控制器，时分复用数字滤波器和信号处理微计算机。 对于数字信号的每个样本，序列控制器被编程为对数字滤波器进行时间复用以对六个音调信号中的每一个执行三个级联二阶滤波操作（两个带通滤波器操作和一个低通滤波器操作），以提供相应的能量估计 以及一个额外的滤波操作来提供总能量估计。 信号处理微处理器处理多个七个能量估计组，并且当检测到多频调色剂对时提供指示。 数字滤波器在通过序列控制器的滤波器起始信号使能时，异步地执行单个类似乘法的滤波操作来实现每个二阶滤波器，并且在完成滤波操作时提供滤波器完成信号。 通过逻辑互补数字滤波器输入信号，在低通滤波操作期间提供全波整流能力。 数字滤波器输出信号的限制周期可以通过对输出信号进行舍入，并将正和负的溢出分别钳位到最大允许的正和负信号。 音调接收机可以有利地用于PCM通信系统中，用于检测用于拨号和监督控制的多频音调信令。 此外，本发明的音调调谐器可以适于仅通过在其中更改固件来接收许多不同类型的音调信号。
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10.发明授权Method of storing reflection coeffients in a vector quantizer for a speech coder to provide reduced storage requirements 失效将反射系数存储在用于语音编码器的矢量量化器中以提供减少的存储要求的方法
公开(公告)号：US5826224A
公开(公告)日：1998-10-20
申请号：US609027
申请日：1996-02-29
IPC分类号： G10L19/06 , G10L19/083 , G10L19/135 , G10L25/06 , G10L25/24 , G10L9/14 , G10L9/08
CPC分类号： G10L19/083 , G10L19/06 , G10L19/135 , G10L25/06 , G10L25/24
摘要： An input speech signal is encoded as one or more reflection coefficients. To reduce storage requirements, the reflection coefficients are scalar quantized by storing an N-bit code rather than the entire reflection coefficient. An exemplary value for N is 8. A table is provided having 2.sup.N reflection coefficient values. The N-bit code is used to look up reflection coefficient values from the table. To reduce spectral distortion due to scalar quantization, the reflection coefficient values in the table are non-linearly scaled.
摘要翻译： 输入语音信号被编码为一个或多个反射系数。 为了减少存储要求，通过存储N位代码而不是整个反射系数来对反射系数进行标量化。 N的示例值为8.提供具有2N个反射系数值的表。 N位代码用于从表中查找反射系数值。 为了减少由于标量量化引起的频谱失真，表中的反射系数值是非线性比例的。
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