公开(公告)号：US5890126A

公开(公告)日：1999-03-30

申请号：US815318

申请日：1997-03-10

申请人： Eric Lindemann

发明人： Eric Lindemann

IPC分类号： G10L7/06

CPC分类号： G10L21/003 ,  G10H7/04

摘要： Apparatus for simultaneously decompressing and interpolating compressed audio data. The compressed audio data is stored in differential log format, meaning that the difference between each two consecutive data points is taken and the log of the difference calculated to form each compressed data point. To efficiently decompress and interpolate the compressed data, advantage is taken of the fact that addition of logs is equivalent to multiplication of linear values. Thus the log of an interpolation factor is added to each compressed data point prior to taking the inverse log of the sum. An integrator block completes the interpolation and decompression of the data.

摘要翻译： 用于同时解压缩和内插压缩音频数据的装置。 压缩音频数据以差分日志格式存储，这意味着采用每两个连续数据点之间的差异，并计算差异日志以形成每个压缩数据点。 为了有效地解压缩和内插压缩数据，优点在于添加日志等价于线性值的乘法。 因此，在获取总和的逆对数之前，将内插因子的对数添加到每个压缩数据点。 积分器块完成数据的插值和解压缩。

公开(公告)号：US6026359A

公开(公告)日：2000-02-15

申请号：US929879

申请日：1997-09-15

申请人： Yoshikazu Yamaguchi ,  Shigeki Sagayama ,  Jun-ichi Takahashi ,  Satoshi Takahashi

发明人： Yoshikazu Yamaguchi ,  Shigeki Sagayama ,  Jun-ichi Takahashi ,  Satoshi Takahashi

IPC分类号： G10L15/06 ,  G10L15/065 ,  G10L15/14 ,  G10L7/06

CPC分类号： G10L15/065 ,  G10L15/144 ,  G10L2015/0635

摘要： A model adaptation scheme in the pattern recognition, which is capable of realizing a fast, real time model adaptation and improving the recognition performance. This model adaptation scheme determines a change in a parameter expressing a condition of pattern recognition and probabilistic model training between an initial condition at a time of acquiring training data used in obtaining a model parameter of each probabilistic model and a current condition at a time of actual recognition. Then, the probabilistic models are adapted by obtaining a model parameter after a condition change by updating a model parameter before a condition change according to the determined change, when the initial condition and the current condition are mismatching. The adaptation processing uses a Taylor expansion expressing a change in the model parameter in terms of a change in the parameter expressing the condition.

摘要翻译： 模式识别中的模型适应方案，能够实现快速，实时的模型适应和提高识别性能。 该模型适应方案确定在获取用于获得每个概率模型的模型参数的实际数据与实际时间的当前条件之间的初始条件时，表示模式识别和概率模型训练条件的参数的变化 承认。 然后，当初始条件和当前条件不匹配时，通过根据所确定的变化更新条件改变之前的模型参数，通过在条件改变之后获得模型参数来适应概率模型。 根据表示条件的参数的变化，适应处理使用表示模型参数的变化的泰勒展开。

公开(公告)号：US5765126A

公开(公告)日：1998-06-09

申请号：US392756

申请日：1995-04-17

申请人： Kyoya Tsutsui ,  Mito Sonohara

发明人： Kyoya Tsutsui ,  Mito Sonohara

IPC分类号： H04B14/04 ,  G10L19/00 ,  G10L19/02 ,  G10L19/06 ,  G10L19/14 ,  H04B1/66 ,  G10L7/06

CPC分类号： G10L19/06 ,  G10L19/002 ,  G10L19/012 ,  G10L19/0212 ,  G10L19/16 ,  H04B1/667 ,  G10L25/18

摘要： A signal encoding apparatus for encoding an acoustic signal. This signal encoding apparatus includes a transform circuit for transforming an inputted acoustic signal into frequency components, a signal component separating circuit for separating an output of the transform circuit into tone characteristic components and noise characteristic components, a tone characteristic encoding circuit for encoding a signal of tone characteristic components, and a noise characteristic component encoding circuit for encoding a signal of noise characteristic components, wherein the tone characteristic component encoding circuit encodes respective signal components of the signal of tone characteristic components so that they respectively have different code lengths to thereby improve efficiency of encoding without degrading sound quality with respect to acoustic signal of tone characteristic.

摘要翻译： PCT No.PCT / JP94 / 01056 Sec。 371日期1995年04月17日 102（e）日期1995年04月17日PCT 1994年6月29日PCTA用于对声信号进行编码的信号编码装置。 该信号编码装置包括用于将输入的声信号变换为频率分量的变换电路，将变换电路的输出分离为音调特性分量和噪声特性分量的信号分量分离电路，用于对变换电路的输出进行编码的色调特性编码电路， 音调特征分量，以及噪声特性分量编码电路，用于对噪声特征分量的信号进行编码，其中，音调特征分量编码电路对音调特征分量的信号的各个信号分量进行编码，使得它们分别具有不同的码长，从而提高效率 相对于音调特性的声信号而言，音质不降低。

公开(公告)号：US5381512A

公开(公告)日：1995-01-10

申请号：US903729

申请日：1992-06-24

申请人： Thomas Holton ,  Steven D. Love ,  Stephen P. Gill

发明人： Thomas Holton ,  Steven D. Love ,  Stephen P. Gill

IPC分类号： G10L15/02 ,  G10L15/18 ,  G10L17/00 ,  G10L5/00 ,  G10L7/06 ,  G10L9/10

CPC分类号： G10L15/02 ,  G10L17/02

摘要： A stimulus waveform is processed using a model of the human auditory system to provide a plurality of output waveforms. Each output waveform corresponds to excitation at different locations along the basilar membrane in the cochlea, and matches the narrow frequency bandwidth, short time response, and wave propagation characteristics of the human cochlea. Primary feature detection is achieved by comparing response waveforms and their spatial and time derivatives to predetermined stereotypes. Secondary feature detection is achieved by comparing spatial and temporal patterns of primary features with patterns stereotypical of human speech elements.

摘要翻译： 使用人类听觉系统的模型来处理刺激波形以提供多个输出波形。 每个输出波形对应于耳蜗基底膜不同位置处的激发，匹配人耳蜗的窄频带宽，短时间响应和波传播特性。 通过将响应波形及其空间和时间导数与预定的刻板印象进行比较来实现主要特征检测。 通过将主要特征的空间和时间模式与人类语音元素的模式刻板比较来实现次要特征检测。

公开(公告)号：US5890106A

公开(公告)日：1999-03-30

申请号：US821017

申请日：1997-03-19

申请人： Marina Bosi-Goldberg

发明人： Marina Bosi-Goldberg

IPC分类号： H03H17/02 ,  G10L7/06

CPC分类号： H03H17/0266

摘要： An efficient implementation of oddly-stacked critically-sampled single sideband analysis/synthesis filter banks is achieved by application of a set of functions to time-domain and frequency-domain values before and after transformation. In one embodiment of an analysis filter bank, a forward pre-transform function groups blocks of N samples into blocks of 1/4N modified samples, a discrete transform generates frequency-domain coefficients in response to the modified samples, and a forward post-transform function generates spectral information in response to the frequency-domain transform coefficients. In one embodiment of a synthesis filter bank, an inverse pre-transform function groups spectral information into blocks of 1/4N frequency-domain transform coefficients, a discrete transform generates blocks of 1/4N time-domain transform coefficients in response to the frequency-domain transform coefficients, and an inverse post-transform function generates blocks of N time-domain samples in response to the time-domain transform coefficients. An implementation of an oddly-stacked Time Domain Aliasing Cancellation transform permits the length of the transformation to be adaptively selected.

摘要翻译： 通过在变换前后对时域和频域值应用一组函数来实现奇数堆叠临界采样单边带分析/合成滤波器组的有效实现。 在分析滤波器组的一个实施例中，正向预变换功能将N个采样的块分组为+ E，fra 1/4 + EE N个修改采样的块，离散变换响应于修改的采样生成频域系数 并且前向后变换功能响应于频域变换系数产生频谱信息。 在合成滤波器组的一个实施例中，逆预变换功能将频谱信息分组为+ E，fra 1/4 + EE N个频域变换系数的块，离散变换生成+ E， + EE N时域变换系数，并且逆后变换函数响应于时域变换系数生成N个时域样本的块。 奇异堆叠的时域混叠取消变换的实现允许自适应地选择变换的长度。

公开(公告)号：US5367608A

公开(公告)日：1994-11-22

申请号：US144092

申请日：1993-10-27

申请人： Raymond N. J. Veldhuis ,  Gerrit J. Keesman

发明人： Raymond N. J. Veldhuis ,  Gerrit J. Keesman

IPC分类号： G06T9/00 ,  H04B1/66 ,  G10L7/06 ,  G10L5/00

CPC分类号： H04B1/667 ,  G06T9/007

摘要： Transmitter, encoding system and method for subband coding a digital signal. The encoding system includes a splitter unit for dividing the digital signal into subband signals SB.sub.1, . . . , SB.sub.p ; a quantizer unit for quantizing time-equivalent q sample signal blocks of the subband signals; a bit need determining unit and a bit allocation unit. The bit need determining unit determines a bit need b.sub.m which corresponds to the number of bits by which the q samples in a time-equivalent signal block in a subband signal SB.sub.m should be represented, where 1.ltoreq.m.ltoreq.P. The bit allocation unit allocates n.sub.m bits to each of the q samples of the time-equivalent signal block of subband signal SB.sub.m on the basis of the bit need b.sub.m and an available bit quantity B, n.sub.m being the number of bits by which the q samples in the time-equivalent signal block of subband signal SB.sub.m will be represented, where 1.ltoreq.m.ltoreq.P.

摘要翻译： 用于子带编码数字信号的发射机，编码系统和方法。 编码系统包括用于将数字信号划分为子带信号SB1的分离器单元。 。 。 ，SBp; 量化器单元，用于量化子带信号的时间等效q采样信号块; 有点需要确定单元和位分配单元。 位需要确定单元确定对应于应当表示子带信号SBm中的时间等效信号块中的q个样本的比特数的比特需求bm，其中1 = P。 比特分配单元基于比特需要bm和可用比特量B将子比特分配给子带信号SBm的时间等效信号块的q个样本中的每一个，其中，q是q个样本的比特数 在子带信号SBm的时间等效信号块中将被表示，其中1≤n≤P。

公开(公告)号：US5347611A

公开(公告)日：1994-09-13

申请号：US822316

申请日：1992-01-17

申请人： Hyokang Chang

发明人： Hyokang Chang

IPC分类号： G10L11/02 ,  H04Q1/453 ,  G10L7/00 ,  G10L7/06

CPC分类号： G10L25/78 ,  H04Q1/453

摘要： A method of passing tones transparently over narrowband channels without having to detect the tones at the transmit side is described. This method requires neither separate handling of tones for the transmission purpose, nor prior knowledge of the characteristics of the signaling tones used. It is a relatively general method that can be applied to systems using a variety of speech coding techniques utilizing a frame structure.

摘要翻译： 描述了在窄带信道上透明地传递音调而不必在发送侧检测音调的方法。 该方法既不需要单独处理用于传输目的的音调，也不需要使用所使用的信令音的特征的先前知识。 这是一种相对一般的方法，可以应用于利用帧结构的各种语音编码技术的系统。

公开(公告)号：USRE36478E

公开(公告)日：1999-12-28

申请号：US631222

申请日：1996-04-12

申请人： Robert J. McAulay ,  Thomas F. Quatieri, Jr.

发明人： Robert J. McAulay ,  Thomas F. Quatieri, Jr.

IPC分类号： G10L19/02 ,  G10L5/00 ,  G10L3/00 ,  G10L7/06

CPC分类号： G10L19/02

摘要： A sinusoidal model for acoustic waveforms is applied to develop a new analysis/synthesis technique which characterizes a waveform by the amplitudes, frequencies, and phases of component sine waves. These parameters are estimated from a short-time Fourier transform. Rapid changes in the highly-resolved spectral components are tracked using the concept of "birth" and "death" of the underlying sine waves. The component values are interpolated from one frame to the next to yield a representation that is applied to a sine wave generator. The resulting synthetic waveform preserves the general waveform shape and is perceptually indistinguishable from the original. Furthermore, in the presence of noise the perceptual characteristics of the waveform as well as the noise are maintained. The method and devices are particularly useful in speech coding, time-scale modification, frequency scale modification and pitch modification.

摘要翻译： 应用用于声波形的正弦模型来开发新的分析/合成技术，其通过分量正弦波的幅度，频率和相位来表征波形。 这些参数是从短时傅里叶变换估计的。 高分辨率光谱分量的快速变化使用基础正弦波“出生”和“死亡”的概念进行跟踪。 分量值从一帧内插到下一帧以产生应用于正弦波发生器的表示。 所得到的合成波形保留了一般的波形形状，并且在感觉上与原始波形不可区分。 此外，在存在噪声的情况下，维持波形的感知特性以及噪声。 该方法和装置在语音编码，时间尺度修改，频率规模修改和音高修改中特别有用。

公开(公告)号：US5677984A

公开(公告)日：1997-10-14

申请号：US392482

申请日：1995-02-23

申请人： Yukio Mitome

发明人： Yukio Mitome

IPC分类号： G01R23/16 ,  G10L11/00 ,  G10L7/06

CPC分类号： G10L25/48 ,  G01R23/16 ,  G10L25/24

摘要： Respective samples of input speech data are transformed by a discrete Fourier transformer to obtain a spectrum of the speech data. Simultaneously, values of times of the respective samples are multiplied with the input speech data by a multiplier and a differential spectrum is obtained by transforming a result of the multiplication by a discrete Fourier transformer. A real part of a value obtained by dividing the differential spectrum by the spectrum by a quotient real part calculator and the real part is inverse-transformed by a discrete inverse Fourier transformer. The result of the inverse-transformation is divided by the values of the times of the respective samples to obtain a time function corresponding to phase. On the other hand, a time function corresponding to a logarithmic amplitude spectrum is obtained from an output of the inverse-Fourier transformer by means of a logarithmic amplitude spectrum calculator. A complex cepstrum is obtained by adding the both time functions at respective times by an adder.

摘要翻译： 输入语音数据的各样本由离散傅立叶变换器变换以获得语音数据的频谱。 同时，通过乘法器将各个样本的时间值与输入的语音数据相乘，并且通过将离散傅里叶变换器的乘法结果变换来获得差分频谱。 通过商实数部分计算器和实部将差分光谱除以光谱获得的值的实部由逆离散付里叶变换器进行逆变换。 将逆变换的结果除以各个样本的时间值，以获得对应于相位的时间函数。 另一方面，通过对数幅度频谱计算器从傅立叶逆变换器的输出获得与对数振幅谱相对应的时间函数。 通过加法器在各自的时间加上两个时间函数来获得复数倒谱。

公开(公告)号：US5394508A

公开(公告)日：1995-02-28

申请号：US822247

申请日：1992-01-17

申请人： Jae S. Lim

发明人： Jae S. Lim

IPC分类号： G10L19/02 ,  G10L21/02 ,  H04B1/66 ,  G10L3/02 ,  G10L7/06

CPC分类号： G10L19/0212 ,  H04B1/667 ,  G10L19/0204 ,  G10L2021/02163 ,  G10L25/27

摘要： An audio type signal is encoded. The signal is first divided into bands. For each band, a yardstick signal element is selected. Its magnitude is quantized using a first level of accuracy. This magnitude is used for various purposes, including assigning bits to the different bands, and for establishing reconstruction levels within a band. The magnitude of non yardstick signal elements is quantized with less accuracy than are the yardstick signal elements. The encoded signal is also decoded. Apparatus for both encoding and decoding are also disclosed. The location of the yardstick element within its band may also be recorded and encoded, and used for efficiently allocating bits to non-yardstick signal elements.

摘要翻译： 音频类型信号被编码。 该信号首先分为频带。 对于每个频带，选择尺码信号元素。 其幅度使用第一级精度进行量化。 该幅度用于各种目的，包括将比特分配给不同的频带，以及用于在频带内建立重建级别。 非标尺信号元素的大小以比标尺信号元素更低的精度进行量化。 编码信号也被解码。 还公开了用于编码和解码的装置。 标尺元件在其频带内的位置也可以被记录和编码，并且用于有效地将比特分配给非标尺信号元素。