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公开(公告)号：US4853958A
公开(公告)日：1989-08-01
申请号：US201154
申请日：1988-06-02
IPC分类号： H04Q1/457
CPC分类号： H04Q1/4575
摘要： A dual tone multi-frequency (DTMF) receiver for use in detecting digitally transmitted signals in the telephone industry. The receiver derives linear predictive coefficients for the digital signals in the data frame. This information is used to compute frequency response magnitudes at the DTMF frequencies. A plurality of magnitude comparisons are then performed to verify the presence of true DTMF signals and concurrently the frequency of these signals is obtained.
摘要翻译： 一种双音多频（DTMF）接收机，用于检测电话行业中的数字传输信号。 接收器为数据帧中的数字信号导出线性预测系数。 该信息用于计算DTMF频率的频率响应幅度。 然后执行多个幅度比较以验证真实DTMF信号的存在，同时获得这些信号的频率。
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2.发明授权
公开(公告)号：US5995923A
公开(公告)日：1999-11-30
申请号：US883353
申请日：1997-06-26
CPC分类号： G10L19/16 , H04W88/181
摘要： In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.
摘要翻译： 近年来，电讯业目睹了各种数字声码器的扩散，以满足不同有线和无线通讯系统的带宽需求。 网络的多样性的快速增长和这种网络的用户数量正在增加两个声码器串联放置以服务于单个连接的情况的数量。 低比特率编解码器的这种布置可能会降低发送语音的质量。 为了克服这个问题，本发明提供了一种用于在无线通信环境中发送数字化语音信号的新颖方法和装置。 该装置能够通过中间公共格式将压缩的语音信号从一种格式转换成另一格式，从而避免必须将语音数据连续解压缩到PCM型数字化，然后再重新压缩语音数据。
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公开(公告)号：US5455888A
公开(公告)日：1995-10-03
申请号：US985418
申请日：1992-12-04
CPC分类号： G10L21/038 , G10L19/0204 , G10L21/0232
摘要： A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.
摘要翻译： 语音带宽扩展方法和装置分析使用LPC分析以8kHz采样的窄带语音，以确定其频谱形状和反向滤波以提取其激励信号。 激励信号内插到16 kHz的采样率，并分析了音调控制和功率电平。 然后对产生白噪声的宽带信号进行滤波以提供合成的宽带激励信号。 确定窄带形状并将其与各矢量量化器码本中的模板进行比较，以选择各自的高频带形状和增益。 然后对合成的宽带激励信号进行滤波以提供高频带信号，该高频带信号又被加到窄带信号中，以16kHz采样率内插，以产生人造宽带信号。 该装置可以在数字信号处理器芯片上实现。
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公开(公告)号：US06975666B2
公开(公告)日：2005-12-13
申请号：US09742421
申请日：2000-12-22
申请人： Sofiène Affes , Henrik Hansen , Paul Mermelstein
发明人： Sofiène Affes , Henrik Hansen , Paul Mermelstein
CPC分类号： H04B1/71052 , H04B7/086
摘要： A receiver of the present invention addresses the need for improved interference suppression without the number of transmissions by the power control system being increased, and, to this end, provides a receiver for a CDMA communications system which employs interference subspace rejection to tune a substantially null response to interference components from selected signals of other user stations. Preferably, the receiver also tunes a substantially unity response for a propagation channel via which a corresponding user's signal was received. The receiver may be used in a base station or in a user/mobile station.
摘要翻译： 本发明的接收机解决了在不增加功率控制系统的传输次数的情况下对改进的干扰抑制的需要，并且为此，为CDMA通信系统提供接收机，该接收机采用干扰子空间抑制来调谐基本上为零 响应来自其他用户站的选定信号的干扰分量。 优选地，接收机还为传播信道调谐基本上一致的响应，通过该传播信道接收相应的用户信号。 接收机可以在基站中或在用户/移动台中使用。
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5.发明授权Apparatus and method for coding speech signals by making use of an adaptive codebook 有权通过使用自适应码本对语音信号进行编码的装置和方法公开(公告)号：US06345255B1
公开(公告)日：2002-02-05
申请号：US09621959
申请日：2000-07-21
申请人： Paul Mermelstein
发明人： Paul Mermelstein
IPC分类号： G10L1900
CPC分类号： G10L19/18
摘要： An audio signal encoding device is provided including an input for receiving a sub-frame of an audio signal to be encoded, an adaptive codebook and a processing unit. The adaptive codebook stores at least one prior knowledge entry which includes a data element representative of characteristics of at least a portion of a previously generated audio signal sub-frame. The processing unit generates a set of parameters allowing for synthesization of the audio signal sub-frame received at the input on the basis of at least the sub-frame of the audio signal received at the input and the data element stored in the adaptive codebook. A corresponding decoding device for synthesizing an audio signal on the basis of a set of parameters is also provided.
摘要翻译： 提供一种音频信号编码装置，包括用于接收要编码的音频信号的子帧的输入，自适应码本和处理单元。 自适应码本存储至少一个先验知识条目，其包括表示先前产生的音频信号子帧的至少一部分的特性的数据元素。 处理单元生成一组参数，其允许基于至少在输入端接收的音频信号的子帧和存储在自适应码本中的数据元素在输入处接收到的音频信号子帧进行合成。 还提供了一种用于基于一组参数来合成音频信号的相应解码装置。
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公开(公告)号：US4956865A
公开(公告)日：1990-09-11
申请号：US191824
申请日：1988-05-02
摘要： In a speech recognizer, for recognizing unknown utterances in isolated-word speech or continuous speech, improved recognition accuracy is obtained by augmenting the usual spectral representation of the unknown utterance with a dynamic component. A corresponding dynamic component is provided in the templates with which the spectral representation of the utterance is compared. In preferred embodiments, the representation is mel-based cepstral and the dynamic components comprise vector differences between pairs of primary cepstra. Preferably the time interval between each pair is about 50 milliseconds. It is also preferable to compute a dynamic perceptual loudness component along with the dynamic parameters.
摘要翻译： 在语音识别器中，为了识别孤立词语音或连续语音中的未知语音，通过用动态分量增加未知语音的常规频谱表示来获得改进的识别精度。 在模板中提供相应的动态分量，与之对比发音的频谱表示。 在优选实施例中，该表示是基于mel的倒频谱，并且动态分量包括主要cepstra对之间的矢量差异。 优选地，每对之间的时间间隔约为50毫秒。 还优选地计算动态感知响度分量以及动态参数。
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公开(公告)号：US5913187A
公开(公告)日：1999-06-15
申请号：US920724
申请日：1997-08-29
申请人： Paul Mermelstein
发明人： Paul Mermelstein
CPC分类号： G10L21/0208 , G10L21/0364 , G10L19/04
摘要： The invention relates to a linear prediction audio signal processing apparatus, such as a vocoder, including a nonlinear filter to attenuate the residual signal used to excite a linear prediction synthesis filter. The nonlinear filter is capable of reducing the noise component in the signal while keeping only the periodic component of the speech signal. This feature enhances speech quality. The invention also extends to a novel method for processing a residual signal used to excite a linear prediction synthesis filter in order to attenuate wide band additive noise in the speech signal as constructed by the synthesis filter.
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8.发明授权Reducing search complexity for code-excited linear prediction (CELP) coding 失效减少代码激励线性预测（CELP）编码的搜索复杂度
公开(公告)号：US5526464A
公开(公告)日：1996-06-11
申请号：US53754
申请日：1993-04-29
申请人： Paul Mermelstein
发明人： Paul Mermelstein
CPC分类号： G10L19/12
摘要： A code-excited linear prediction (CELP) coding method and code divide the residual signal into frequency bands. Codebooks provided for each band decrease in size with increasing band frequency. Reduction in codebook size with increasing frequency together with reduction in sampling rate with decreasing frequency provide reductions in codebook search complexity that allow real time implementation on digital signal processor chips.
摘要翻译： 码激励线性预测（CELP）编码方法和码将剩余信号划分为频带。 为每个频带提供的码本随着频带频率的增加而减小。 随着频率的增加，码本尺寸的减小以及采样频率随频率的降低而降低，从而降低了码本搜索的复杂度，从而实现数字信号处理器芯片的实时性。
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9.发明授权Apparatus and method for coding speech signals by making use of voice/unvoiced characteristics of the speech signals 失效通过利用语音信号的语音/清音特征对语音信号进行编码的装置和方法公开(公告)号：US06249758B1
公开(公告)日：2001-06-19
申请号：US09107385
申请日：1998-06-30
申请人： Paul Mermelstein
发明人： Paul Mermelstein
IPC分类号： G10L1900
CPC分类号： G10L19/18
摘要： An audio signal encoding device is provided comprising an input for receiving a sub-frame of an audio signal, a voiced audio signal synthesis stage, an unvoiced audio signal synthesis stage, and a processing unit. The voiced audio signal synthesis stage is operative for producing a first synthetic audio signal approximating the sub-frame of an audio signal received at the input on the basis of a first set of parameters. The unvoiced audio signal synthesis stage is operative for producing a second synthetic audio signal approximating the sub-frame of an audio signal received at the input on the basis of a second set of parameters. The processing unit is operative for releasing a set of parameters allowing to generate a selected one of the first synthetic audio signal and the second synthetic audio signal.
摘要翻译： 提供了一种音频信号编码装置，包括用于接收音频信号的子帧的输入，有声音频信号合成级，无声音频信号合成级和处理单元。 有声音频信号合成级用于产生基于第一组参数来逼近在输入端接收的音频信号的子帧的第一合成音频信号。 无声音频信号合成级用于产生基于第二组参数近似在输入端接收的音频信号的子帧的第二合成音频信号。 处理单元用于释放允许生成第一合成音频信号和第二合成音频信号中所选择的一个参数的一组参数。
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10.发明授权Nonlinear filter for noise suppression in linear prediction speech processing devices 失效线性预测语音处理设备中噪声抑制的非线性滤波器
公开(公告)号：US6052659A
公开(公告)日：2000-04-18
申请号：US289970
申请日：1999-04-13
申请人： Paul Mermelstein
发明人： Paul Mermelstein
CPC分类号： G10L21/0208 , G10L21/0364 , G10L19/04
摘要： The invention relates to a linear prediction audio signal processing apparatus, such as a vocoder, including a nonlinear filter to attenuate the residual signal used to excite a linear prediction synthesis filter. The nonlinear filter is capable of reducing the noise component in the signal while keeping only the periodic component of the speech signal. This feature enhances speech quality. The invention also extends to a novel method for processing a residual signal used to excite a linear prediction synthesis filter in order to attenuate wide band additive noise in the speech signal as constructed by the synthesis filter.
摘要翻译： 本发明涉及一种线性预测音频信号处理装置，诸如声码器，包括用于衰减用于激励线性预测合成滤波器的残余信号的非线性滤波器。 非线性滤波器能够降低信号中的噪声分量，同时只保留语音信号的周期分量。 该功能增强了语音质量。 本发明还延伸到一种用于处理用于激励线性预测合成滤波器的残余信号以便衰减由合成滤波器构成的语音信号中的宽带附加噪声的新颖方法。
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