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1. 发明授权

US06408267B1 Method for decoding an audio signal with correction of transmission errors 有权
标题翻译：用传输错误校正来解码音频信号的方法
公开(公告)号：US06408267B1
公开(公告)日：2002-06-18
申请号：US09402529
申请日：2000-01-14
申请人： Stéphane Proust
发明人： Stéphane Proust
IPC分类号： G10L1904
CPC分类号： G10L19/005
摘要： The decoder receives a bit stream representative of an audio signal, with a flag indicating any missing frames. For each frame, an excitation signal is formed from excitation parameters recovered in the bit stream if the frame is valid and estimated otherwise if the frame is missing, and the excitation signal is filtered by means of a synthesis filter to obtain a decoded audio signal. A linear prediction analysis is performed on the basis of the decoded audio signal obtained up to the preceding frame to estimate at least in part a synthesis filter relating to the current frame, whereby the successive synthesis filters used to filter the excitation signal, as long as there is no missing frame, conform to the estimated synthesis filters. If a frame n0 is missing, at least one synthesis filter used to filter the excitation signal relative to a subsequent frame n0+i is determined by a weighted combination of the synthesis filter estimated in relation to frame n0+i and at least one synthesis filter that has been used since frame n0.
摘要翻译：解码器接收表示音频信号的比特流，其中标志指示任何丢失的帧。对于每个帧，如果帧是有效的，则在比特流中恢复的激励参数形成激励信号，否则如果帧丢失则估计激励信号，并且通过合成滤波器对激励信号进行滤波以获得解码的音频信号。基于直到前一帧获得的解码音频信号来执行线性预测分析，至少部分地估计与当前帧相关的合成滤波器，由此用于过滤激励信号的连续合成滤波器，只要没有丢失帧，符合估计的合成滤波器。如果帧n0丢失，则相对于后续帧n0 + i滤波激励信号的至少一个合成滤波器是通过相对于帧n0 + i估计的合成滤波器和至少一个合成滤波器的加权组合来确定的这是从第n0帧起使用的。

2. 发明授权

US06353808B1 Apparatus and method for encoding a signal as well as apparatus and method for decoding a signal 有权
标题翻译：用于编码信号的装置和方法以及用于对信号进行解码的装置和方法
公开(公告)号：US06353808B1
公开(公告)日：2002-03-05
申请号：US09422250
申请日：1999-10-21
申请人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
发明人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
IPC分类号： G10L1904
CPC分类号： G10L19/0212 , G10L19/0204 , G10L19/09
摘要： An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit. A coefficient quantization section quantizes the coefficients from the orthogonal transform circuit section according to the number of allocated bits from the bit allocation calculation section.
摘要翻译：通过正交变换在时基上对输入信号进行编码的装置和方法包括基于通过线性预测编码（LPC）分析获得的参数和对输入信号的音调分析来去除信号波形的相关性在正交变换之前的时基。来自输入端子的时基输入信号被发送到归一化电路部分和LPC分析电路。归一化电路部分去除信号波形的相关性，并通过LPC逆滤波器和音调反向滤波器取出残差，并将其发送到正交变换电路部分。来自LPC分析电路的LPC参数和来自音调分析电路的音调参数被发送到比特分配计算电路。系数量化部根据来自比特分配计算部的分配比特数，对来自正交变换电路部的系数进行量化。

3. 发明授权

US06339757B1 Bit allocation method for digital audio signals 失效
标题翻译：数字音频信号的位分配方法
公开(公告)号：US06339757B1
公开(公告)日：2002-01-15
申请号：US08161798
申请日：1993-12-06
申请人： Do Hui Teh , Pek Yew Tan , Sua Hong Neo
发明人： Do Hui Teh , Pek Yew Tan , Sua Hong Neo
IPC分类号： G10L1904
CPC分类号： H04B1/665
摘要： When determining a quantization for digital audio signals having a spectral and temporal structure wherein the audio signals are buffered in frames and decomposed into spectral components, the method is: obtaining a variance and/or a representative of the components of the the frame within a defined frequency interval; computing a necessary bandwidth using psychoacoustic criteria; determining an initial quantization for each frequency interval using an approximate mathematical model; and iteratively increasing the quantization of each of the frequency intervals through a psychoacoustically weighted inverse relationship with the the variance of each of the frequency intervals.
摘要翻译：当确定具有频谱和时间结构的数字音频信号的量化时，其中音频信号被缓冲在帧中并且被分解为频谱分量，该方法是：在所定义的帧内获得方差和/或代表帧的分量频率间隔; 使用心理声学标准计算必要的带宽; 使用近似数学模型确定每个频率间隔的初始量化; 并且通过与每个频率间隔的方差的心理声学加权的反相关关系迭代地增加每个频率间隔的量化。

4. 发明授权

US06304843B1 Method and apparatus for reconstructing a linear prediction filter excitation signal 有权
标题翻译：用于重建线性预测滤波器激励信号的方法和装置
公开(公告)号：US06304843B1
公开(公告)日：2001-10-16
申请号：US09226914
申请日：1999-01-05
申请人： Hung-Bun Choi , Harvey Hau-Fai Wong , Wing Tak Kenneth Wong
发明人： Hung-Bun Choi , Harvey Hau-Fai Wong , Wing Tak Kenneth Wong
IPC分类号： G10L1904
CPC分类号： G10L19/08
摘要： An apparatus and method of reconstructing a linear prediction synthesis filter excitation signal, by: receiving a signal representative of output from a linear prediction synthesis filter, producing therefrom a deterministic signal comprising a magnitude spectrum (50) and a phase spectrum (52); and producing (54) the reconstructed excitation signal from the deterministic signal and a noise signal.
摘要翻译：一种通过以下方式重建线性预测合成滤波器激励信号的装置和方法：接收表示来自线性预测合成滤波器的输出的信号，从而产生包括幅度谱（50）和相位谱（52）的确定性信号; 以及从确定性信号和噪声信号产生（54）重建的激励信号。

5. 发明授权

US06240382B1 Efficient codebook structure for code excited linear prediction coding 失效
标题翻译：用于码激励线性预测编码的高效码本结构
公开(公告)号：US06240382B1
公开(公告)日：2001-05-29
申请号：US08734356
申请日：1996-10-21
申请人： Daniel Lin
发明人： Daniel Lin
IPC分类号： G10L1904
CPC分类号： G10L19/12 , G10L2019/0005 , G10L2019/0007
摘要： A speech communication system using a code excited linear prediction speech decoder. The decoder using a first codebook containing a first digital value sequence selected from the set of binary values {0, 1}. The decoder also using a second codebook containing a second digital value sequence having values selected from the set of binary values {−1, 0}. The first digital value sequence and the second digital value sequence are combined to become a third digital value sequence having a set of ternary values from the set of {−1, 0, 1}.
摘要翻译：一种使用码激励线性预测语音解码器的语音通信系统。该解码器使用包含从二进制值{0,1}组中选择的第一数字值序列的第一码本。解码器还使用包含具有从二进制值{-1,0}组中选择的值的第二数字值序列的第二码本。第一数字值序列和第二数字值序列被组合成具有来自{-1,0,1}的集合的具有三元值的集合的第三数字值序列。

6. 发明授权

US06223157B1 Method for direct recognition of encoded speech data 失效
标题翻译：用于直接识别编码语音数据的方法
公开(公告)号：US06223157B1
公开(公告)日：2001-04-24
申请号：US09074726
申请日：1998-05-07
申请人： Thomas D. Fisher , Jeffery J. Spiess , Dearborn R. Mowry
发明人： Thomas D. Fisher , Jeffery J. Spiess , Dearborn R. Mowry
IPC分类号： G10L1904
CPC分类号： G10L15/30 , G10L15/02
摘要： Digital Cellular telephony requires voice compression designed to minimize the bandwidth required for the digital cellular channel. The features used in speech recognition have similar components to those used in the vocoding process. The present invention provides a system that bypasses the de-compression or decoding phase of the vocoding and converts the digital cellular parameters directly into features that can be processed by a recognition engine. More specifically, the present invention provides a system and method for mapping a vocoded representation of parameters defining speech components, which in turn define a particular waveform, into a base feature type representation of parameters defining speech components (e.g. LPC parameters), which in turn define the same digital waveform.
摘要翻译：数字蜂窝电话需要语音压缩，旨在最小化数字蜂窝信道所需的带宽。语音识别中使用的特征与声码过程中使用的特征具有相似的组成部分。本发明提供了一种绕过声码的去压缩或解码阶段的系统，并将数字蜂窝参数直接转换成可由识别引擎处理的特征。更具体地说，本发明提供一种系统和方法，用于将定义语音分量的参数的语音代码表示（其又将特定波形）映射成定义语音分量（例如LPC参数）的参数的基本特征类型表示，其依次定义相同的数字波形。

7. 发明授权

US06192334B1 Audio encoding apparatus and audio decoding apparatus for encoding in multiple stages a multi-pulse signal 失效
标题翻译：用于多级编码多脉冲信号的音频编码装置和音频解码装置
公开(公告)号：US06192334B1
公开(公告)日：2001-02-20
申请号：US09053606
申请日：1998-04-01
申请人： Toshiyuki Nomura
发明人： Toshiyuki Nomura
IPC分类号： G10L1904
CPC分类号： G10L19/18 , G10L19/10 , G10L19/107
摘要： Auxiliary multi-pulse setting circuit 130 set candidates of pulse positions so that the pulse positions to which no pulse is located are selected in auxiliary multi-pulse searching circuit 131 prior to the pulse positions at which pulses have already been encoded in multi-pulse searching circuit 110. Auxiliary multi-pulse searching circuit 131 generates an auxiliary multi-pulse signal according to the candidates of pulse positions set in auxiliary multi-pulse setting circuit 130 and encodes the auxiliary multi-pulse signal so that difference between the reproduced audio signal which is obtained by driving a linear predictive synthesis filter with the auxiliary multi-pulse signal and an input audio signal is minimized similarly to multi-pulse searching circuit 110.
摘要翻译：辅助多脉冲设置电路130设置脉冲位置的候选，使得在多脉冲搜索中脉冲已被编码的脉冲位置之前在辅助多脉冲搜索电路131中选择没有脉冲位置的脉冲位置辅助多脉冲搜索电路131根据在辅助多脉冲设置电路130中设置的脉冲位置的候选产生辅助多脉冲信号，并且对辅助多脉冲信号进行编码，使得再现的音频信号之间的差异通过用辅助多脉冲信号驱动线性预测合成滤波器并且类似于多脉冲搜索电路110最小化输入音频信号而获得。

8. 发明授权

US06757650B2 Excitation vector generator, speech coder and speech decoder 有权
标题翻译：激励矢量发生器，语音编码器和语音解码器
公开(公告)号：US06757650B2
公开(公告)日：2004-06-29
申请号：US09855708
申请日：2001-05-16
申请人： Kazutoshi Yasunaga , Toshiyuki Morii , Taisuke Watanabe , Hiroyuki Ehara
发明人： Kazutoshi Yasunaga , Toshiyuki Morii , Taisuke Watanabe , Hiroyuki Ehara
IPC分类号： G10L1904
CPC分类号： G10L19/135 , G10L19/12 , G10L2019/0007 , G10L2019/0013
摘要： A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.
摘要翻译：传统CELP型语音编码器/解码器的随机码矢量读取部分和随机码本分别被替换为根据输入种子的值输出不同矢量流的振荡器和用于存储多个种子的种子存储部分。这使得不必像固定码本（ROM）一样存储固定向量，从而显着地降低了存储器容量。

9. 发明授权

US06732069B1 Linear predictive analysis-by-synthesis encoding method and encoder 有权
标题翻译：线性预测分析合成编码方法和编码器
公开(公告)号：US06732069B1
公开(公告)日：2004-05-04
申请号：US09396300
申请日：1999-09-15
申请人： Erik Ekudden , Roar Hagen
发明人： Erik Ekudden , Roar Hagen
IPC分类号： G10L1904
CPC分类号： G10L19/083
摘要： A linear predictive analysis-by-synthesis encoder includes a search algorithm block (50) and a vector quantizer (58) for vector quantizing optimal gains from a plurality of subframes in a frame. The internal encoder states are updated (50, 52, 54, 56) using the vector quantized gains.
摘要翻译：线性预测分析合成编码器包括搜索算法块（50）和矢量量化器（58），用于矢量量化来自帧中的多个子帧的最佳增益。使用向量量化增益更新内部编码器状态（50,52,54,56）。

10. 发明授权

US06654716B2 Perceptually improved enhancement of encoded acoustic signals 有权
标题翻译：感知改善了编码声信号的增强
公开(公告)号：US06654716B2
公开(公告)日：2003-11-25
申请号：US09982029
申请日：2001-10-19
申请人： Stefan Bruhn , Susanne Olvenstam
发明人： Stefan Bruhn , Susanne Olvenstam
IPC分类号： G10L1904
CPC分类号： G10L21/038 , G10L19/02
摘要： The invention relates to encoding of broadband and narrowband acoustic source signals (x) such that the perceived sound quality of corresponding reconstructed signals is improved in comparison to the known solutions. An enhancement estimation unit (102), operating in serial or in parallel with the regular encoding/decoding means (101), perceptually enhances a reconstructed acoustic source signal by utilization of an enhancement spectrum (C) comprising a larger number of spectral coefficients than the number of sample values in corresponding frames of the signals carrying the basic encoded representation of the acoustic source signal. The thus extended block length of the enhancement spectrum frame provides a basis for accomplishing the desired improvement of the perceived sound quality.
摘要翻译：本发明涉及宽带和窄带声源信号（x）的编码，使得与已知解决方案相比，相应重构信号的感知音质得到改善。与常规编码/解码装置（101）串联或并行操作的增强估计单元（102）通过利用包括比所述编码/解码装置（101）更大数量的频谱系数的增强频谱（C）感知地增强重构声源信号携带声源信号的基本编码表示的信号的相应帧中的采样值的数量。增强频谱框的扩展块长度为实现感知音质的期望改善提供了基础。

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