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1.发明授权Method for coding/decoding, coding/decoding device, and videoconferencing apparatus using such device 失效使用这种装置的编码/解码方法，编码/解码装置和视频会议装置
公开(公告)号：US5983172A
公开(公告)日：1999-11-09
申请号：US759085
申请日：1996-11-29
申请人： Makoto Takashima , Yoshiaki Asakawa
发明人： Makoto Takashima , Yoshiaki Asakawa
IPC分类号： G01L9/00 , G10L11/00 , G10L19/00 , H03M7/30 , H04N19/115 , H04N19/124 , H04N19/126 , H04N19/136 , H04N19/147 , H04N19/169 , H04N19/176 , H04N19/189 , H04N19/196 , H04N19/423 , H04N19/50 , H04N19/59 , H04N19/60 , H04N19/61 , H04N19/625 , H04N19/85 , H04N19/94
CPC分类号： G10L19/022 , G10L19/002 , G10L19/032
摘要： The object of the invention is to provide a coding/decoding method in which degradation of sound quality perceptible by the listener does not occur at an low bit rate. A shift number calculation section of a decoding device divides a frequency domain into at least two sub-bands, and approximates each of normalized transform coefficients in the sub-band whose allocated bit value is less than a predetermined threshold using a quantized value of the transform coefficient in a predetermined sub-band other than the sub-band so as to obtain information concerning the approximation, and a multiplexer multiplexes the information and another signal and transmits them. A de-multiplexer of a decoding device separates the code of information concerning the approximation, and a shift number restore section restores the information based thereon. An approximation coefficient calculation section assigns, based on the information concerning the approximation, the transform coefficient values in the predetermined sub-band to the normalized transform coefficients whose allocated bit value is less than the predetermined threshold.
摘要翻译： 本发明的目的是提供一种编码/解码方法，其中以低比特率不会发生收听者感知到的声音质量的劣化。 解码装置的移位数计算部分将频域划分为至少两个子带，并且使用变换的量化值近似分配的比特值小于预定阈值的子带中的每个归一化变换系数 系数，以获得关于该近似的信息，并且复用器复用该信息和另一个信号并将其发送。 解码装置的去多路复用器分离有关近似的信息代码，并且移位数恢复部分基于此恢复信息。 近似系数计算部分根据关于近似的信息将预定子带中的变换系数值分配给其分配比特值小于预定阈值的归一化变换系数。
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公开(公告)号：US5956686A
公开(公告)日：1999-09-21
申请号：US497474
申请日：1995-06-30
CPC分类号： G10L19/0212
摘要： An adaptive transform coding/and decoding arrangement is provided to effectively exploit different redundancies between the bands of a spectrum envelope to effect coding at a low bit rate for an audio signal. In the adaptive transform coding method, the spectrum envelope is divided into bands so that different coding methods may be applied to the spectrum envelopes of the individual bands. By applying the present invention to the adaptive transform coding of an audio signal, the spectrum envelope can be adjusted to the coding/and transmission method which is suitable for the time fluctuation in each frequency band, so that the different redundancies for the individual bands can be effectively exploited to realize a highly efficient audio signal coding/and decoding method which has its bits reduced as required for coding the spectrum envelope.
摘要翻译： 提供自适应变换编码/解码装置以有效地利用频谱包络的频带之间的不同冗余度来实现音频信号的低比特率的编码。 在自适应变换编码方法中，将频谱包络分为频带，使得不同的编码方法可以应用于各个频带的频谱包络。 通过将本发明应用于音频信号的自适应变换编码，可以将频谱包络调整为适合于每个频带中的时间波动的编码和/或传输方法，使得各个频带的不同冗余可以 有效地实现高效率的音频信号编码/解码方法，其对于对频谱包络进行编码所需的位减少。
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公开(公告)号：US4985923A
公开(公告)日：1991-01-15
申请号：US328702
申请日：1989-03-27
申请人： Akira Ichikawa , Yoshiaki Asakawa , Akio Komatsu , Eiji Oohira
发明人： Akira Ichikawa , Yoshiaki Asakawa , Akio Komatsu , Eiji Oohira
摘要： A voice coding system for separating and coding voice information into spectrum envelope information and voice source information, with the intention of compressing the amount of information for efficient coding of vocal audio signals through the control of the voice source information based on the fact that the spectrum envelope information and voice source information highly correlate with each other.
摘要翻译： 一种用于将语音信息分离和编码成频谱包络信息和语音源信息的语音编码系统，其目的是通过基于以下事实来压缩用于语音源信息的有效编码的信息量：频谱 信封信息和声源信息彼此高度相关。
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公开(公告)号：US4975957A
公开(公告)日：1990-12-04
申请号：US343892
申请日：1989-04-24
CPC分类号： G10L19/038 , G10L19/0018 , G10L19/06
摘要： A character voice communication system including high efficiency voice coding system for encoding and transmitting speech information at a high efficiency and a voice character input/output system for converting speech information into character information or receiving character information and transmitting speech or character information are organically integrated. A speech analyzer and a speech synthesizer are shared by both the voice coding and the voice character input/output systems. Communication apparatus is also provided which allows mutual conversion between speech signals and character codes.
摘要翻译： 包括用于高效率地编码和发送语音信息的高效语音编码系统和用于将语音信息转换为字符信息或接收字符信息并传送语音或字符信息的语音字符输入/输出系统的字符语音通信系统被有机地整合。 语音分析器和语音合成器由语音编码和语音字符输入/输出系统共享。 还提供了允许语音信号和字符代码之间的相互转换的通信装置。
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5.发明授权
公开(公告)号：US5142582A
公开(公告)日：1992-08-25
申请号：US511768
申请日：1990-04-20
CPC分类号： H04B1/66
摘要： In speech decoding, a transmission code, which includes an error correcting code added to a speech code, is received and whether or not there is a code error is detected on the basis of the error correcting code. At this time, when there is no code error or when the detected code error has been corrected, a normal speech decoding processing is executed. On the other hand, when there is a code error which is impossible to be corrected, artificially background sound corresponding to the decoded speech is generated from characteristic parameters indicating unvoiced sound in the decoded speech. The parameters are continuously extracted from the decoded speech, stored in a memory and are used to replace an erroneous portion of the speech code.
摘要翻译： 在语音解码中，接收包括添加到语音码的纠错码的发送码，并根据纠错码检测是否存在码错误。 此时，当没有代码错误或者当检测到的代码错误已经被校正时，执行正常语音解码处理。 另一方面，当存在不可能校正的代码错误时，从解码语音中指示无声音的特征参数产生与解码语音相对应的人为背景音。 这些参数从被解码的语音中连续提取，存储在存储器中，并用于替换语音码的错误部分。
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公开(公告)号：US4528688A
公开(公告)日：1985-07-09
申请号：US601957
申请日：1984-04-19
申请人： Akira Ichikawa , Akio Komatsu , Yoshiaki Asakawa
发明人： Akira Ichikawa , Akio Komatsu , Yoshiaki Asakawa
CPC分类号： G10L15/00
摘要： This speech signal recognition system compares the two-dimensionals pattern (time sequence of feature vectors) of an unknown signal to prestored standard references patterns for recognition, thus forming a corresponding two-dimensional comparison pattern of points of elemental Hamming distance differences. The sum of the pattern point distances is the similarity measure. To improve accuracy, partial patterns are selected (or "masked") and tested sequentially, and the point values weighted relative to their location within the mask. The mask may be rectangular or oblique.
摘要翻译： 该语音信号识别系统将未知信号的二维图案（特征向量的时间序列）与预先存储的标准参考图案进行比较，从而形成元素汉明距离差的点的对应二维比较模式。 图案点距离的总和是相似性度量。 为了提高准确性，选择（或“屏蔽”）部分图案并进行顺序测试，并且点值相对于其在掩模内的位置进行加权。 面具可以是矩形或斜面。
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公开(公告)号：US5119424A
公开(公告)日：1992-06-02
申请号：US282497
申请日：1988-12-12
CPC分类号： G10L19/10
摘要： A speech signal is analyzed for each frame so that it is separated into spectral envelope information and excitation information, and the excitation information is expressed by a plurality of pulses. Judgement is conducted as to whether the current frame is a voiced frame immediately after the transition from an unvoiced frame, a voiced frame continuative from a voiced frame or an unvoiced frame, and excitation pulses are generated in accordance with the judgement result. In case of a continuing voiced frame, the excitation pulse position of the current voiced frame is determined based on the pitch period with respect to the excitation pulse position of the immediately preceding voiced frame so that the excitation pulse train is generated at a position approximated to the determined position.
摘要翻译： 针对每个帧分析语音信号，使其被分离为频谱包络信息和激励信息，并且激励信息由多个脉冲表示。 进行关于当前帧是否是从无声帧过渡之后的有声帧，从有声帧或无声帧连续的有声帧，以及根据判断结果生成激励脉冲的判断。 在连续有声帧的情况下，基于相对于紧接在前的有声帧的激励脉冲位置的音调周期来确定当前有声帧的激励脉冲位置，使得激励脉冲串在近似于 确定的位置。
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公开(公告)号：US5077798A
公开(公告)日：1991-12-31
申请号：US412987
申请日：1989-09-26
IPC分类号： G06T9/00 , G10L19/038 , H03M7/30
CPC分类号： G10L19/038 , G06T9/008 , H03M7/3082 , Y10S706/90
摘要： A system for voice coding based on vector quantization has an apparatus in which a distribution area of parameters representative of a voice is divided into a plurality of domains so that one vector (code vector) may correspond to one domain, an apparatus for representing individual code vectors by codes specific thereto, an apparatus for converting an input voice into a vector and determining membership functions by numerically expressing the distance between the nearest code vector and each of the predetermined number of neighboring vectors, and an apparatus for transmitting, as fuzzy vector quantization information, a code of the nearest code vector and the membership functions.
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公开(公告)号：US4776015A
公开(公告)日：1988-10-04
申请号：US804938
申请日：1985-12-05
申请人： Shoichi Takeda , Akira Ichikawa , Yoshiaki Asakawa
发明人： Shoichi Takeda , Akira Ichikawa , Yoshiaki Asakawa
CPC分类号： G10L19/10
摘要： Herein disclosed is a speech analysis-synthesis apparatus which resorts to a multi-pulse exciting method using a plurality of modeled pulses as a synthetic sound source if input speech is analyzed so that speech may be synthesized on the basis of the analyzed result. A factor for effecting perpetual weighting in a manner to correspond to the sound source pulse number is made variable, and the error between the input speech and the synthesized speech is perceptually weighted so that the amplitude and location of the train of the sound source pulses are so determined as to minimize said error.
摘要翻译： 这里公开了一种语音分析合成装置，如果分析输入语音，则使用多个建模脉冲作为合成声源的多脉冲激励方法，以便可以基于分析结果来合成语音。 以与声源脉冲数对应的方式进行永久加权的因素变为可变，并且输入语音和合成语音之间的误差被感知加权，使得声源脉冲的列的幅度和位置为 所以确定为最小化所述错误。
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公开(公告)号：US4718095A
公开(公告)日：1988-01-05
申请号：US554960
申请日：1983-11-25
摘要： A speech recognition method makes it possible to improve the accuracy of recognition of input speech and is capable of operating on a real time basis. This is accomplished by generating from the input speech signal a difference signal which indicates whether the speech power of the input speech is increasing or decreasing for each frame. The similarity between the input speech and a standard pattern is then calculated for each frame, and this is then followed by correcting the similarity calculation on the basis of the generated difference signal and a difference signal relating to the standard pattern obtained from storage. The matching of the input speech and the standard pattern is then effected by using the corrected similarity, and the input speech is then recognized from the result of this matching. Thus, a spectrum matching distance weighted by power information of speech can be obtained in real time.
摘要翻译： 语音识别方法使得可以提高输入语音的识别精度并能够实时地进行操作。 这是通过从输入语音信号生成指示输入语音的语音功率对于每个帧是增加还是减少的差分信号来实现的。 然后针对每个帧计算输入语音与标准模式之间的相似度，然后根据生成的差分信号和与从存储获得的标准模式相关的差分信号来校正相似度计算。 然后通过使用校正的相似度来实现输入语音和标准模式的匹配，然后从该匹配的结果中识别输入语音。 因此，可以实时获得通过语音功率信息加权的频谱匹配距离。
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