公开(公告)号：US07873110B2

公开(公告)日：2011-01-18

申请号：US10463315

申请日：2003-06-17

申请人： Ravindra Bidnur ,  Girish Hulmani ,  Manoj Kumar Vajhallya

发明人： Ravindra Bidnur ,  Girish Hulmani ,  Manoj Kumar Vajhallya

IPC分类号： H04N7/12 ,  G10L3/02

CPC分类号： H04N21/4341 ,  H04N21/23406 ,  H04N21/235 ,  H04N21/4348 ,  H04N21/435 ,  H04N21/44004

摘要： An interrupt sensitive extract byte instruction scheme is presented herein. The interrupt sensitive extract by instruction extracts bytes from data, depending on the presence of an interrupt. The extract byte instruction extracts bytes from data in the absence of the interrupt and does not extract bytes in the presence of the interrupt. The interrupt can be triggered by a set of counters that count the number of extracted bytes. By loading the counters with a particular number, the interrupt can be generated when the particular number of data bytes is extracted.

摘要翻译： 本文介绍了中断敏感提取字节指令方案。 中断敏感提取按指令从数据中提取字节，具体取决于是否存在中断。 提取字节指令在没有中断的情况下从数据中提取字节，并且在存在中断的情况下不会提取字节。 中断可以由一组计数提取字节数的计数器触发。 通过加载具有特定数字的计数器，可以在提取特定数量的字节数时产生中断。

公开(公告)号：US6049771A

公开(公告)日：2000-04-11

申请号：US985395

申请日：1997-12-05

申请人： Isao Yamamoto

发明人： Isao Yamamoto

IPC分类号： G10L19/00 ,  G11B20/00 ,  G11C7/16 ,  H03M7/38 ,  G10L3/02

CPC分类号： G11C7/16 ,  G11B20/00007

摘要： An apparatus carries out an adaptive pulse code modulation process on voice data and records them in a solid memory which is detachably attached to the apparatus. The code bit number is switched, depending on the selected recording mode. A filter for suppressing a high-voice region of the frequency band of the voice data is used, and there is provided a selector for applying this filter at the time of voice reproduction selectively either always or according to the code bit number.

摘要翻译： 一种装置对语音数据进行自适应脉冲编码调制处理，并将其记录在可拆卸地附接到装置上的固态存储器中。 根据所选的录制模式，切换码位号。 使用用于抑制语音数据的频带的高语音区域的滤波器，并且提供选择器，用于在语音再现时总是或根据码位数来应用该滤波器。

公开(公告)号：US6041297A

公开(公告)日：2000-03-21

申请号：US814130

申请日：1997-03-10

申请人： Randy G. Goldberg

发明人： Randy G. Goldberg

IPC分类号： G10L19/12 ,  G10L3/02 ,  G10L9/00

CPC分类号： G10L19/12

摘要： A vocoder according to the present invention includes an analyzer portion and a synthesizer portion. The analyzer portion encodes an input frame of speech on the basis of a candidate excitation selected from a group of candidate excitations stored in memory. Instead of transmitting the actual candidate excitation to the synthesizer portion, the analyzer portion generates and provides to the synthesizer portion a variable length index code that identifies the selected candidate excitation. The synthesizer portion stores in memory the same plurality of candidate excitations as the analyzer portion. The synthesizer portion uses the variable length index code to obtain from its memory the candidate excitation originally selected by the analyzer portion. The synthesizer portion reconstructs the input frame of speech on the basis of the obtained candidate excitation.

摘要翻译： 根据本发明的声码器包括分析器部分和合成器部分。 分析器部分基于从存储在存储器中的一组候选激励中选择的候选激励来对输入的语音帧进行编码。 分析器部分不是向合成器部分发送实际的候选激励，而是向合成器部分生成标识选择的候选激励的可变长度索引代码。 合成器部分在存储器中存储与分析器部分相同的多个候选激励。 合成器部分使用可变长度索引代码从其存储器获得原始由分析器部分选择的候选激励。 合成器部分基于获得的候选激励重建输入的语音帧。

公开(公告)号：US6014617A

公开(公告)日：2000-01-11

申请号：US905545

申请日：1997-08-04

申请人： Hideki Kawahara

发明人： Hideki Kawahara

IPC分类号： G01R23/02 ,  G10L25/90 ,  G10L3/02

CPC分类号： G10L25/90

摘要： A speech signal input from a microphone is distributed by a distribution amplifier. Using output signals of a filter group of cos phase having cut-off frequency moderate on low frequency side and steep on high frequency side and of similar filter group of sin phase, stability index is calculated based on magnitude of amplitude modulation and magnitude of frequency modulation of the signals, by stability index calculating portion and fundamental frequency extracting portion. Based on the result of calculation, approximate value of fundamental frequency is calculated based on an output of a channel indicating maximum stability, and based on the approximate value of fundamental frequency, instantaneous frequency extracting portion extracts precise instantaneous frequency as fundamental frequency, interpolating value of instantaneous frequency from adjacent frequency channels.

摘要翻译： 从麦克风输入的语音信号由分配放大器分配。 在低频侧，陡峭的高频侧和类似滤波器组的sin相中，使用具有中频偏置频率的滤波器组的输出信号，基于幅度调制幅度和频率调制幅度来计算稳定性指标 的信号，由稳定度指标计算部分和基频提取部分。 基于计算结果，基于表示最大稳定性的信道的输出计算基频的近似值，并且基于基频的近似值，瞬时频率提取部分将精确瞬时频率提取为基频，内插值 来自相邻频道的瞬时频率。

公开(公告)号：US6009387A

公开(公告)日：1999-12-28

申请号：US821747

申请日：1997-03-20

申请人： Ganesh Nachiappa Ramaswamy ,  Ponani Gopalakrishnan ,  Joseph Morris

发明人： Ganesh Nachiappa Ramaswamy ,  Ponani Gopalakrishnan ,  Joseph Morris

IPC分类号： G10L19/00 ,  G10L19/04 ,  G10L3/02

CPC分类号： G10L19/04

摘要： Apparatus for processing acoustic features extracted from a sample of speech data forming a feature vector signal every frame period includes a first linear prediction analyzer, a vector quantizer, at least one partitioned vector quantizer and a scalar quantizer. The first linear prediction analyzer performs a linear prediction analysis on the feature vector signal to generate a first error vector signal. Next, the vector quantizer performs a vector quantization on the first error signal thereby generating a first index corresponding to a first prestored vector signal which is an approximation of the first error vector signal. The vector quantizer also generates a residual vector signal which is the difference between the first error vector signal and the first prestored approximation vector signal. Next, the at least one partitioned vector quantizer performs a partitioned vector quantization on a first portion of the residual vector signal thereby generating at least one second index corresponding to a second prestored vector signal which is an approximation of the first portion of the residual vector signal. Next, the scalar quantizer performs a scalar quantization on a second portion of the residual vector signal thereby generating a third index corresponding to a prestored scalar signal which is an approximation of the second portion of the residual vector signal. The first, second and third indices are combined to form an encoded vector signal which is a compressed representation of the feature vector signal. The encoded vector signal may be transmitted and/or stored as desired. The feature vector signal may be reconstructed from the encoded vector signal by adding the corresponding prestored signals to the encoded vector signal to form a decompressed representation of the feature vector signal.

摘要翻译： 用于处理从每帧周期形成特征向量信号的语音数据样本提取的声学特征的装置包括第一线性预测分析器，矢量量化器，至少一个分割矢量量化器和标量量化器。 第一线性预测分析器对特征矢量信号进行线性预测分析，生成第一误差矢量信号。 接下来，矢量量化器对第一误差信号执行矢量量化，从而生成与作为第一误差矢量信号的近似的第一预存储矢量信号相对应的第一索引。 矢量量化器还产生残差矢量信号，其是第一误差矢量信号和第一预先存储的近似矢量信号之间的差。 接下来，至少一个分割矢量量化器对残差矢量信号的第一部分执行分割矢量量化，从而生成对应于第二预存矢量信号的至少一个第二索引，其是残差向量信号的第一部分的近似值 。 接下来，标量量化器对剩余矢量信号的第二部分执行标量量化，从而生成对应于作为剩余矢量信号的第二部分的近似的预存储标量信号的第三索引。 第一，第二和第三索引被组合以形成编码矢量信号，其是特征向量信号的压缩表示。 可以根据需要发送和/或存储经编码的矢量信号。 可以通过将对应的预存储信号加到编码矢量信号中，从编码矢量信号重构特征向量信号，以形成特征向量信号的解压缩表示。

公开(公告)号：US5991724A

公开(公告)日：1999-11-23

申请号：US35106

申请日：1998-03-05

申请人： Hideki Kojima ,  Shinta Kimura

发明人： Hideki Kojima ,  Shinta Kimura

IPC分类号： G11B20/10 ,  G10L13/08 ,  G10L21/04 ,  G10L3/02

CPC分类号： G10L21/04

摘要： A reproduction speed of speech sound changing apparatus which reproduces speech data at a speed in which essential part thereof can be caught so that the outline of the speech sound can be grasped even when changing the reproduction speed, besides remarkably reduces the whole reproducing time wherein a reproducing speed in each predetermined period is calculated according to a parameter value in every predetermined period of speech data in accordance with such a manner that a part having a high parameter value such as high power, high pitch or the like of speech data is judged to be the part, where important contents are involved, and such part of important contents is reproduced at such a speed that the contents can be caught, while the parts other than that described above are reproduced either at such a speed that the whole reproduction of speech data can be completed within a required time, or reproduced by skipping over the parts if at thus determined reproduction speed, reproduced speech sound cannot be caught, as a result of paying attention to such fact that voice is louder or pitch of voice becomes higher in the part containing important contents in speech data.

摘要翻译： 语音改变装置的再现速度以能够捕获其主要部分的速度再现语音数据，即使在改变再现速度时也可以掌握语音的轮廓，同时显着地减少了整个再现时间，其中a 根据每个预定的语音数据周期中的参数值，按照以下方式计算每个预定周期中的再现速度，即，将具有诸如语音数据的高功率，高音调等的高参数值的部分判断为 作为涉及重要内容的部分，并且以可以捕获内容的速度再现这些重要内容的部分，而除了上述内容之外的部分以使得语音的整个再现的速度再现 数据可以在所需时间内完成，或者如果以如此确定的再现速度再现，则通过跳过部件来再现 由于在语音数据中包含重要内容的部分中声音更大或语音音调更高的事实，因此不能捕获语音。

公开(公告)号：US5983192A

公开(公告)日：1999-11-09

申请号：US234856

申请日：1999-01-22

申请人： Stephen C. Botzko ,  David M. Franklin

发明人： Stephen C. Botzko ,  David M. Franklin

IPC分类号： H04L12/56 ,  G10L11/02 ,  G10L19/00 ,  H04B1/20 ,  H04M3/56 ,  H04M11/00 ,  H04R3/00 ,  G10L3/02 ,  H04L12/16

CPC分类号： H04M3/568 ,  G10L19/00 ,  H04B1/207 ,  H04M3/56 ,  G10L2025/783

摘要： A method and apparatus for distributing audio signals from one of a plurality of audio sources to an output connect compressed audio signals from each one of the plurality of audio sources to an audio processor. Uncompressed audio signals are derived from the compressed audio signals. The compressed audio signal from one of the plurality of audio sources is selectively coupled to the output based upon the uncompressed audio signals. In a preferred embodiment, the compressed audio signal from one of the plurality of audio sources is coupled to the output selectively in accordance with speech information detected in the uncompressed audio signals from the plurality of audio sources. The method and apparatus include mixing, selectively, the uncompressed audio signals from the plurality of audio sources into a composite uncompressed audio signal; compressing the uncompressed composite signal into a corresponding compressed composite signal; and connecting the compressed composite signal or the selected one of the compressed audio signals to the output selectively in accordance with the uncompressed audio signals. In a preferred embodiment of the invention, the compressed composite signal is fed to the output when speech audio signals are detected from more than one source and the selected one of the compressed audio signals is coupled to the output when speech is detected from only one of the sources.

公开(公告)号：US5978760A

公开(公告)日：1999-11-02

申请号：US897852

申请日：1997-07-21

申请人： Ajit V. Rao ,  Wilfrid P. LeBlanc

发明人： Ajit V. Rao ,  Wilfrid P. LeBlanc

IPC分类号： G10L19/00 ,  H03M7/30 ,  H04B14/00 ,  H04B15/00 ,  G10L3/02

CPC分类号： G10L19/012

摘要： To overcome the problem of poor representation of the background noise, the present invention includes a noise parameter generator (40) which uses a weighted average of auto-correlation values of the input signal generated during the noise-analysis phase. The weighting function gives less weight to the auto-correlations during the first few frames (as they may contain speech) and more weight to frames towards the end of this phase. Also included, to overcome the bursty nature of comfort noise, is a comfort noise generator (50) which gradually changes the nature of the signal from speech to pseudo-random noise after the speech-burst The comfort noise generator (50) of the present invention excites the auto-regressive filter corresponding to the noise model with a weighted combination of the past excitation and pseudo-random noise.

摘要翻译： 为了克服背景噪声的表现差的问题，本发明包括使用在噪声分析阶段期间产生的输入信号的自相关值的加权平均值的噪声参数发生器（40）。 加权函数对最初几帧（因为它们可能包含语音）的自相关给予较小的权重，并且在该阶段结束时对帧的权重更大。 还包括为了克服舒适噪声的突发性质，是一种舒适噪声发生器（50），其在语音突发之后逐渐地将信号的性质从语音改变为伪随机噪声。本发明的舒适噪声发生器（50） 本发明利用过去的激励和伪随机噪声的加权组合来激励对应于噪声模型的自回归滤波器。

公开(公告)号：US5978759A

公开(公告)日：1999-11-02

申请号：US157419

申请日：1998-09-21

申请人： Mineo Tsushima ,  Yoshihisa Nakatoh ,  Takeshi Norimatsu

发明人： Mineo Tsushima ,  Yoshihisa Nakatoh ,  Takeshi Norimatsu

IPC分类号： G10L21/0232 ,  G10L21/038 ,  G10L25/12 ,  G10L3/02

CPC分类号： G10L21/038 ,  G10L21/0232 ,  G10L25/12

摘要： Apparatus for expanding the bandwidth of speech signals such that a narrowband speech signal is input and digitized, the spectral envelope information and residual information are extracted from the digitized signal by linear predictive coding analysis, the spectral envelope information is expanded into wideband information by a spectral envelope converter, the residual information is expanded into wideband information by a residual converter, the converted spectral envelope information and residual information are combined to produce a wideband speech signal, frequency information not contained in the input signal is extracted from the obtained wideband speech signal by a filter, and the resulting signal is added to the original digitized input signal, and the obtained signal is converted into an analog signal as the output signal of the apparatus. The apparatus comprises a linear mapping function codebook used for converting spectral parameters, and a weights calculator and an adder for weighing and summing function outputs.

摘要翻译： 用于扩展语音信号的带宽使得窄带语音信号被输入和数字化的装置，通过线性预测编码分析从数字化信号中提取频谱包络信息和残差信息，频谱包络信息通过频谱扩展成宽带信息 通过残差转换器将残差信息扩展为宽带信息，将经转换的频谱包络信息和残差信息组合起来产生宽带语音信号，从所获得的宽带语音信号中提取不包含在输入信号中的频率信息， 滤波器，并且将所得到的信号加到原始的数字化输入信号上，并将获得的信号转换为模拟信号作为装置的输出信号。 该装置包括用于转换频谱参数的线性映射函数码本，以及权重计算器和用于对功能输出求和的加法器。

公开(公告)号：US5978758A

公开(公告)日：1999-11-02

申请号：US890881

申请日：1997-07-10

申请人： Shigeru Ono

发明人： Shigeru Ono

IPC分类号： G10L15/02 ,  G10L19/00 ,  G10L19/06 ,  H03M7/30 ,  G10L3/02

CPC分类号： G10L19/07 ,  H03M7/3082

摘要： A first vector quantizer generates output codevectors corresponding in number to a number determined by a predetermined number of bits through linear coupling of integer coefficients of a predetermined number of base vectors stored in a base vector memory. A second vector quantizer determines coefficients of the base vectors according to at least one of output indexes of the output codevectors.

摘要翻译： 第一矢量量化器通过线性耦合存储在基矢量存储器中的预定数量的基矢量的整数系数来生成数量对应于由预定位数确定的数字的输出码矢量。 第二矢量量化器根据输出码矢量的输出索引中的至少一个来确定基矢量的系数。