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公开(公告)号：US4053712A
公开(公告)日：1977-10-11
申请号：US717732
申请日：1976-08-24
申请人： Adolf Reindl
发明人： Adolf Reindl
CPC分类号： H04B1/66
摘要： An adaptive coder and decoder for reducing the bit rate required to trans digitally encoded speech signals. The invention relies on the fact that the speech pattern of the average talker contains significant numbers of inter-syllable and inter-word pauses. The coder includes circuitry that monitors the idle pattern code generated by the coder's analog-to-digital converter and a code word generator that generates one or more special code words that are substituted for idle pattern code sequences of predetermined length. The speech signal and special code words are then fed into an elastic buffer and transmitted to the distant receiver at a lower bit rate than was employed for the encoding process. At the receiving location, the decoder recognizes the special code words, substitutes an idle pattern bit stream of the appropriate length for the code words and then reads-out the contents of an elastic input buffer at a higher bit rate than was used to transmit the incoming signal, that is, at the same bit rate as was initially used at the coder.
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公开(公告)号：US3715512A
公开(公告)日：1973-02-06
申请号：US3715512D
申请日：1971-12-20
发明人： KELLY J
CPC分类号： G10L19/04
摘要： In an adaptive, predictive coder for speech signals, the transmitted signal generally consists of an rms value, a pitch signal, a voice-unvoiced indication, and a number of parameter signals for adjusting the coefficients of a linear predictor. Transmission of these signals is improved in this invention by generating a low rate pulsive signal and by shaping its spectrum in accordance with the parameter signals. The pulsive signals thus act as a carrier for the parameters. The bandwidth required for transmitting the resulting composite signal, i.e., the modulated pulsive signal and the subsidiary signals, is substantially less than that of the original speech signal and somewhat less than that required for the transmission of predictively coded signals.
摘要翻译： 在用于语音信号的自适应预测编码器中，所发送的信号通常由有效值，音调信号，语音清音指示和用于调整线性预测器的系数的参数信号的数量组成。 通过产生低速度脉动信号和根据参数信号整形其频谱，本发明改进了这些信号的传输。 因此，脉动信号用作参数的载体。 传输所得到的复合信号所需的带宽，即调制的脉冲信号和辅助信号，显着小于原始语音信号的带宽，稍微小于传输预测编码信号所需的带宽。
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3.发明授权
公开(公告)号：US5606618A
公开(公告)日：1997-02-25
申请号：US173850
申请日：1993-12-27
申请人： Gerardus C. P. Lokhoff , Y. F. Dehery , G. Stoll , G. Theile
发明人： Gerardus C. P. Lokhoff , Y. F. Dehery , G. Stoll , G. Theile
IPC分类号： G01L7/04 , G10L19/00 , G10L19/02 , G10L19/14 , G11B20/00 , G11B20/10 , G11B20/12 , G11B20/18 , G11B27/30 , H03M5/14 , H04B1/66 , H04B14/04 , H04H20/28 , H04H20/30 , H04H20/31 , H04H20/47 , H04H20/88 , H04H20/89 , H04H20/95 , H04H40/18 , H04H40/36 , H04H60/07 , H04H60/13 , H04H60/18 , H04J3/16 , H04L7/00 , H04N7/08 , H04H5/00 , G10L1/06
CPC分类号： H04H20/88 , G10L19/008 , G10L19/0208 , G11B20/00007 , G11B20/10 , G11B20/10527 , G11B20/12 , G11B20/1833 , G11B27/3027 , H04B1/665 , H04B1/667 , H04H20/28 , H04H20/30 , H04H20/47 , H04H20/89 , H04H20/95 , H04H40/36 , H04H60/07 , H04H60/13 , G11B20/00086 , G11B20/00992 , G11B2020/10592 , G11B2020/10601 , G11B2220/90 , G11B2220/913 , G11B2220/93 , H04H20/31 , H04H40/18 , H04H60/18
摘要： Reproduction accuracy of, for example a digital stereo audio signal, is improved by transmitting sample data as sub-signals such as frequency subband signals. In one or more subbands, corresponding components such as left and right stereo channels are combined so that only one composite signal is transmitted per subband. An indicator signal is transmitted, indicating which subbands are combined. Scale factor signals for all subbands, and for the relative intensity of the respective subband signals which were combined, may also be transmitted. In the receiver a subband signal is derived for each channel from the composite signal, before synthesis of the full channel signals which will be reproduced.
摘要翻译： 通过将样本数据作为子信号（例如频率子带信号）发送来改善例如数字立体声音频信号的再生精度。 在一个或多个子带中，组合诸如左立体声通道和右立体声通道的相应组件，使得每个子频带仅传送一个复合信号。 发送指示符信号，指示哪些子带被组合。 也可以发送所有子带的比例因子信号，以及组合的各个子带信号的相对强度。 在接收机中，在合成将被再现的全信道信号之前，从复合信号中为每个信道导出子带信号。
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公开(公告)号：US3437757A
公开(公告)日：1969-04-08
申请号：US3437757D
申请日：1966-06-15
发明人： COKER CECIL H
CPC分类号： G10L19/02
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5.发明授权
公开(公告)号：US4574392A
公开(公告)日：1986-03-04
申请号：US400958
申请日：1982-07-22
申请人： Ruediger Reiss
发明人： Ruediger Reiss
CPC分类号： G10L19/02
摘要： A synthesizer for a channel vocoder for the transmission of speech with considerable frequency band reduction in digital technology provides a reduction of circuit expense. This is accomplished, given non-recursive filters having finite impulse response, not with a weighting of the pulse-shaped excitation variable with the transmitted envelope values of the spectral channels, but with a time variance of the filters by weighting their filter coefficients with the transmitted envelope values. In addition, with proper dimensioning of the filter coefficients, an optimum speech quality of synthesized speech is obtained at the output of the synthesizer even given elimination of the transmission of a voiced/voiceless signal.
摘要翻译： 用于在数字技术中具有相当大的频带减少的用于语音传输的信道声码器的合成器降低了电路费用。 这是由于给定了具有有限脉冲响应的非递归滤波器，而不是使用具有频谱信道的发射包络值的脉冲激励变量的加权，而是通过对滤波器的时间方差加权， 传输包络值。 另外，通过合适的滤波器系数的大小，即使在消除了有声/无声信号的传输的情况下，也可以在合成器的输出端获得合成语音的最佳语音质量。
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公开(公告)号：US4401855A
公开(公告)日：1983-08-30
申请号：US211115
申请日：1980-11-28
CPC分类号： G10L19/06
摘要： Improved apparatus for the linear predictive coding of human speech in which the speech is sampled through the use of analog filters and the linear predictive coding computations are performed with respect to such samples using digital techniques. The filters are MOS switched capacitor filters which can be implemented on a silicon chip together with the digital circuitry. Specific circuits for implementing two different linear predictive coding speech analysis techniques are disclosed.
摘要翻译： 通过使用模拟滤波器对语音进行线性预测编码的改进的装置，并且使用数字技术对这些样本执行线性预测编码计算。 滤波器是MOS开关电容滤波器，其可以与数字电路一起在硅芯片上实现。 公开了用于实现两种不同的线性预测编码语音分析技术的具体电路。
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7.发明授权Device for converting a non-uniformly sampled signal with short-time spectrum to a uniformly sampled signal 失效用于将具有短时频谱的非均匀采样信号转换成均匀采样信号的装置
公开(公告)号：US4271500A
公开(公告)日：1981-06-02
申请号：US8616
申请日：1979-02-01
申请人： Tore Fjallbrant
发明人： Tore Fjallbrant
摘要： A device for forming omitted samples by extrapolation in a transmitted signal having non-uniform, grouped samples representing a short-time stationary signal such as speech. The grouped samples are passed through a series of delay circuits. Fourier transformation of the samples is performed to form omitted samples which are inserted in the signal between the grouped samples. The Fourier transformation is used to select multiplication coefficients which are multiplied by the sample amplitudes, with the products of the multiplication being summed to form the omitted samples. In one embodiment the extrapolation is performed in both carry forward and carry back circuits.
摘要翻译： 一种用于通过在具有表示短时间固定信号（例如语音）的不均匀的分组样本的发送信号中的外推来形成省略的样本的装置。 分组的样本通过一系列延迟电路。 执行样本的傅立叶变换以形成插入到分组样本之间的信号中的省略的样本。 傅里叶变换用于选择乘以样本幅度的乘法系数，乘法乘积相加以形成省略的样本。 在一个实施例中，外推在进位和回送电路中执行。
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公开(公告)号：US4149039A
公开(公告)日：1979-04-10
申请号：US846832
申请日：1977-10-31
CPC分类号： H04B1/662
摘要： A frequency band compression method, using Time Compression, for use with a frequency division multiplex (FDM) signal carrying voice signal. On the transmission end, the multiplexed signal is divided into a plurality of time segments, each segment having a certain duration, then deleting a part of the each segment and time-expanding the remaining part of the segment by a sawtooth variable delay, thus lowering the frequency of the signal to a predetermined ratio. On the receive end, the signal segments are compressed by a corresponding variable delay, thus making the received frequency higher and reproducing the original multiplexed voice signals.
摘要翻译： 使用时间压缩的频带压缩方法，用于携带语音信号的频分复用（FDM）信号。 在发送端，将多路复用信号分割为多个时间段，每个段具有一定的持续时间，然后删除每个段的一部分，并且通过锯齿可变延迟对段的剩余部分进行时间扩展，从而降低 信号的频率达到预定的比例。 在接收端，信号段被相应的可变延迟压缩，从而使接收的频率更高，并再现原来的多路话音信号。
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公开(公告)号：US3786195A
公开(公告)日：1974-01-15
申请号：US3786195D
申请日：1971-08-13
发明人： SCHIFFMAN M
IPC分类号： G10L21/04 , G11B5/00 , G11B21/00 , H03H7/30 , H03H11/26 , H03K4/502 , H03K7/08 , H04B1/66 , H04B3/10 , G10L1/06
CPC分类号： H04B3/10 , G10L21/04 , G11B5/00 , G11B21/00 , H03H7/30 , H03H11/265 , H03K4/502 , H03K7/08 , H04B1/66 , H04B1/662
摘要： An electrical delay line which has variable delay controlled by a signal input thereto is connected in a sound signal channel for signals such as human speech, to compress or expand the sound signal waveform depending on whether the time delay in the line is increased or decreased. By periodically sweeping the delay line from minimum to maximum time delay or vice-versa, repeated segments of a continuous sound signal waveform are processed so that an output audio signal can be obtained having the original frequency components of the signal and occupying a time duration which is equal to or smaller or larger than the original sound sequence with the successive segments of the signal processed by the variable delay line assembled with regard both to the significant parameters of human speech or other coding and the electrical conditions imposed by the system to produce a composite audio signal which is an intelligible replica of the original and substantially free of annoying aberrations introduced by the delay line processor. Variable delay using analog or digital signal storage is also provided.
摘要翻译： 具有由输入信号控制的可变延迟的电延迟线被连接在用于诸如人类语音的信号的声音信号通道中，以根据线路中的时间延迟是否增加来压缩或扩大声音信号波形。 通过周期性地将延迟线从最小时间扫描到最大时间延迟，反之亦然，对连续声音信号波形的重复段进行处理，使得可以获得具有信号的原始频率分量的输出音频信号，并且占据一段时间 等于或小于或大于原始声音序列，其中由可变延迟线处理的信号的连续片段组合在人类语音或其他编码的重要参数以及系统施加的电气条件下，以产生 复合音频信号，其是原始的可理解的复本，并且基本上没有由延迟线处理器引入的烦人的像差。 还提供了使用模拟或数字信号存储的可变延迟。
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10.发明授权Learning vector quantization and a temporary memory such that the codebook contents are renewed when a first speaker returns 失效学习矢量量化和临时存储器，使得当第一说话者返回时，码本内容被更新
公开(公告)号：US5797118A
公开(公告)日：1998-08-18
申请号：US512311
申请日：1995-08-08
申请人： Akitoshi Saito
发明人： Akitoshi Saito
CPC分类号： G06T9/008 , G10L19/12 , H03M7/3082
摘要： An encoding/decoding system employing vector quantization realizes a high quality encoding and decoding with decreased quantizing errors, employing a small sized codebook which faithfully represents each of the inputted waveform vectors. An encoding/decoding system includes an encoding apparatus and a decoding apparatus, each having a codebook for storing information vectors representative of a predetermined number of signal patterns and index that determine the information vectors. The encoding apparatus compares a vector representing an object signal to be quantized with each information vector in the codebook, selects an information vector that is closest to the vector and outputs an index for the information vector. The decoding apparatus obtains an information vector corresponding to the index obtained at the encoding apparatus side by referring to the codebook and decodes the object signal. The codebook utilizes a temporary memory connected thereto. The content of the codebook is temporarily moved to the temporary memory when the identity of the speaker changes. The contents of the temporary memory are read out when the original speakers returns to the system.
摘要翻译： 采用矢量量化的编码/解码系统采用一种忠实地表示每个输入的波形向量的小尺寸码本，实现具有降低的量化误差的高质量编码和解码。 编码/解码系统包括编码装置和解码装置，每个编码装置和解码装置具有用于存储表示预定数量的信号模式的信息矢量的码本和确定信息矢量的索引。 编码装置将表示要量化的对象信号的矢量与码本中的每个信息矢量进行比较，选择最靠近矢量的信息矢量并输出信息矢量的索引。 解码装置通过参照码本获取与在编码装置一侧获得的索引相对应的信息矢量，对该对象信号进行解码。 码本利用与其连接的临时存储器。 当扬声器的身份改变时，码本的内容临时移动到临时存储器。 当原始扬声器返回到系统时，临时存储器的内容被读出。
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