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21.发明授权Speech decoding method and apparatus to control the reproduction speed by changing the number of transform coefficients 失效语音解码方法和装置，用于通过改变变换系数的数量来控制再现速度
公开(公告)号：US5899966A
公开(公告)日：1999-05-04
申请号：US736211
申请日：1996-10-25
CPC分类号： G10L21/04 , G10L19/0212 , G10L19/12 , G10L25/27
摘要： A signal decoding method and apparatus in which the speech signal reproducing speed is controlled without changing the phoneme or the pitch, in which the apparatus has a data number convertor for converting the number of orthogonal transform coefficients entering a transmission signal input terminal from N to M, an inverse orthogonal transform unit for inverse orthogonal-transforming the M number of the orthogonal transform coefficients obtained by the data number convertor, and a linear predictive coding synthesis filter for performing predictive synthesis based on the short-term prediction residuals obtained by the inverse orthogonal transform unit. For an input signal, short-term prediction residuals are found and are orthogonally transformed to form the orthogonal transform coefficients at a rate of N coefficients per transform unit. The frequency positions of the N transform coefficients may be rearranged to M values by M/N or by oversampling to change N to M. A portable radio terminal embodying the invention is described.
摘要翻译： 一种信号解码方法和装置，其中在不改变音素或音调的情况下控制语音信号再现速度，其中装置具有数据数转换器，用于将进入发送信号输入端的正交变换系数的数目从N转换为M ，用于对由数据数转换器获得的M个正交变换系数进行逆正交变换的逆正交变换单元和用于基于由逆正交获得的短期预测残差执行预测合成的线性预测编码合成滤波器 变换单元。 对于输入信号，发现短期预测残差并且以每变换单位N个系数的速率进行正交变换以形成正交变换系数。 N个变换系数的频率位置可以通过M / N重排为M个值，或者通过过采样将N改变为M.描述体现本发明的便携式无线电终端。
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22.发明授权Continuous and discontinuous sine wave synthesis of speech signals from harmonic data of different pitch periods 失效来自不同音调周期的谐波数据的语音信号的连续和不连续的正弦波合成
公开(公告)号：US5832437A
公开(公告)日：1998-11-03
申请号：US515913
申请日：1995-08-16
申请人： Masayuki Nishiguchi , Jun Matsumoto
发明人： Masayuki Nishiguchi , Jun Matsumoto
CPC分类号： G10L19/02 , G10L19/0204 , G10L25/27
摘要： A method for decoding encoded speech signals uses sine wave synthesis based on harmonics of the original speech signal. The harmonics are obtained by transforming the original speech signal from a time domain to a frequency domain, and the harmonics are arranged as sequential frames with the harmonics of a given frame having a pitch period that may or may not be the same as the pitch period of another frame. According to the decoding method, data arrays respectively containing amplitude data and phase data of the harmonics are zero-padded to provide the arrays with a pre-set number of elements. Inverse orthogonal tarnsformation of the data arrays produces time domain information used to generate a time domain waveform signal for restoring the encoded speech signals. The different pitch periods of the frames are normalized to each other either by smooth (continuous) or acute (discontinuous) interpolation depending on the degree of change in the pitch period between the frames.
摘要翻译： 用于对编码语音信号进行解码的方法使用基于原始语音信号的谐波的正弦波合成。 通过将原始语音信号从时域变换为频域来获得谐波，并且将谐波排列为具有与音调周期相同或可能不相同的音调周期的给定帧的谐波的顺序帧 的另一帧。 根据解码方法，分别包含振幅数据和谐波的相位数据的数据阵列被零填充以向阵列提供预定数量的元素。 数据阵列的正交正交信号产生用于产生用于恢复编码的语音信号的时域波形信号的时域信息。 根据帧之间的音调周期的变化程度，帧的不同音调周期通过平滑（连续）或锐（不连续）插值彼此归一化。
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公开(公告)号：US08364496B2
公开(公告)日：2013-01-29
申请号：US12234768
申请日：2008-09-22
IPC分类号： G10L21/00
CPC分类号： G11B20/00007 , G10L19/24 , G11B20/10527 , G11B20/1251 , G11B2020/00043 , G11B2020/10564 , G11B2020/1289 , G11B2020/1292
摘要： A signal recording and reproducing apparatus includes an encoder encoding an input signal to produce a first group of encoded data, and a second group of encoded data used for reproducing a signal of higher quality than a signal resulting from decoding of the first group of encoded data, a recording unit recording record-data, including the first group and the second group of encoded data, into a recording medium, a reproducing unit reproducing the record-data from the recording medium, a decoder decoding at least the first group of encoded data out of the record-data from the reproducing unit, and a controller controlling an operation of each part of the recording and reproducing apparatus, and the controller performs control so as to cause the recording unit to erase the second group of encoded data according to a command to increase the amount of free storage capacity of the recording medium.
摘要翻译： 信号记录和再现装置包括编码输入信号以产生第一组编码数据的编码器和用于再现比由第一组编码数据解码产生的信号更高质量的信号的第二组编码数据 记录单元将包括第一组和第二组编码数据的记录数据记录到记录介质中，再现单元从记录介质再现记录数据，解码器至少解码第一组编码数据 来自再现单元的记录数据，以及控制记录和再现装置的每个部分的操作的控制器，并且控制器执行控制，以使得记录单元根据一个 命令来增加记录介质的可用存储容量。
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24.发明授权Audio signal processing method, audio signal processing apparatus, Hi-Fi video apparatus, digital video apparatus and 8 mm video apparatus 有权音频信号处理方法，音频信号处理装置，Hi-Fi视频装置，数字视频装置和8mm视频装置公开(公告)号：US06961513B2
公开(公告)日：2005-11-01
申请号：US09734512
申请日：2000-12-11
IPC分类号： G11B5/008 , G11B5/53 , G11B20/10 , G11B20/12 , G11B20/18 , H04N5/782 , H04N5/7826 , H04N5/783 , H04N9/806 , H04N19/89 , H04N7/04 , H04N7/64
CPC分类号： G11B15/026 , G11B5/0086 , G11B5/531 , G11B15/125 , G11B15/1875 , G11B20/10527 , G11B20/1201 , G11B20/18 , G11B20/1876 , G11B2020/10546 , H04N5/78213 , H04N5/78263 , H04N5/783 , H04N9/8063
摘要： An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.
摘要翻译： 一种音频信号处理方法，用于修复诸如噪声，不连续性和中断声音的异常状态，包括检测音频信号的异常状态，删除异常段中的音频信号，通过参考以下内容推导出正确的音频信号 在删除段之前和之后的音频信号的波形，基于推导结果生成用于修复被删除段中的信号的修复信号，将修复信号插入到删除的段中，并将其连接到前后的音频信号 已删除的段。
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公开(公告)号：US20050187775A1
公开(公告)日：2005-08-25
申请号：US11112370
申请日：2005-04-22
IPC分类号： G11B5/008 , G11B5/53 , G11B20/10 , G11B20/12 , G11B20/18 , H04N5/782 , H04N5/7826 , H04N5/783 , H04N9/806 , H04N19/89 , G10L11/00
CPC分类号： G11B15/026 , G11B5/0086 , G11B5/531 , G11B15/125 , G11B15/1875 , G11B20/10527 , G11B20/1201 , G11B20/18 , G11B20/1876 , G11B2020/10546 , H04N5/78213 , H04N5/78263 , H04N5/783 , H04N9/8063
摘要： An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.
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公开(公告)号：US06532443B1
公开(公告)日：2003-03-11
申请号：US08951028
申请日：1997-10-15
IPC分类号： G10L1914
摘要： The processing volume in calculating a weight value for perceptually weighted vector quantization is decreased to speed up the processing or to minimize hardware. To this end, an inverted LPC finds LPC (linear prediction coding) residuals of an input speech signal which are processed with sinusoidal analysis encoding by a sinusoidal analysis encoding unit. The resulting parameters are processed by a vector quantizer with perceptually weighted vector quantization. For this perceptually weighted vector quantization, the weight value is calculated based on results of an orthogonal transform of parameters derived from the impulse response of the transfer function of the weight.
摘要翻译： 在计算感知加权矢量量化的权重值时的处理量减少以加速处理或使硬件最小化。 为此，反相LPC找到用正弦分析编码单元用正弦分析编码处理的输入语音信号的LPC（线性预测编码）残差。 所得到的参数由感知加权矢量量化的矢量量化器处理。 对于这种感知加权矢量量化，基于从权重的传递函数的脉冲响应导出的参数的正交变换的结果来计算权重值。
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27.发明授权Voiced/unvoiced decision using a plurality of sigmoid-transformed parameters for speech coding 失效使用多个S形变换参数进行语音编码的发声/清音决定
公开(公告)号：US06023671A
公开(公告)日：2000-02-08
申请号：US833970
申请日：1997-04-11
CPC分类号： G10L25/93
摘要： A method and apparatus for voiced/unvoiced decision for judging whether an input speech signal is voiced or unvoiced. The input parameters for performing the voiced/unvoiced (V/UV) decision are comprehensively judged in order to enable high-precision V/UV decision by a simplified algorithm. Parameters for the voiced/unvoiced (V/UV) decision include the frame-averaged energy of the input speech signal lev, the normalized autocorrelation peak value r0r, the spectral similarity degree pos, the number of zero crossings nZero, and the pitch lag pch. If these parameters are denoted by x, these parameters are converted by function calculation circuits using a sigmoid function g(x) represented byg(x)=A/(1+exp (-(x-b)/a))where A, a, and b are constants differing with each input parameter. Using the parameters converted by this sigmoid function g(x), the voiced/unvoiced decision is made a V/UV decision circuit.
摘要翻译： 用于用于判断输入语音信号是有声还是无声的有声/无声决定的方法和装置。 综合判断用于执行有声/无声（V / UV）判定的输入参数，以便通过简化算法实现高精度V / UV判定。 有声/无声（V / UV）决定的参数包括输入语音信号lev的帧平均能量，归一化自相关峰值r0r，频谱相似度pos，过零次数nZero和音调滞后pch 。 如果这些参数由x表示，这些参数由函数计算电路使用由g（x）= A /（1 + exp（ - （xb）/ a））表示的S形函数g（x）转换，其中A，a， b是与每个输入参数不同的常数。 使用由该S形函数g（x）转换的参数，将有声/无声决定作为V / UV判定电路。
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公开(公告)号：US5953696A
公开(公告)日：1999-09-14
申请号：US935695
申请日：1997-09-23
申请人： Masayuki Nishiguchi , Jun Matsumoto
发明人： Masayuki Nishiguchi , Jun Matsumoto
CPC分类号： G10L21/0364 , G10L21/0264 , G10L25/15 , H04R2225/43
摘要： Nasalized sound effects during reproduction of low-pitch sounds are suppressed to produce playback sounds of high clarity. Amplitude data is processed with high range formant emphasis of crests and valleys of the envelope of the frequency spectrum on the high frequency range and with deepening of the valley of the frequency spectrum over the entire frequency range, above all, over the low to mid frequency range. Next, the amplitude data is processed for emphasizing the peak values of the formant of the voiced frame in the portion of the speech signal which is rising in magnitude and for unconditionally emphasizing the spectral envelope on the high frequency range. The voiced speech spectrum is generated by synthesizing the cosine wave based upon the emphasized amplitude data.
摘要翻译： 在低声音的再现期间的鼻音化效果被抑制以产生高清晰度的播放声音。 振幅数据以高频范围内的频谱包络的波峰和波谷的高范围共振峰强化处理，并且在整个频率范围内的频谱范围的深度越来越高，尤其是低频到中频 范围。 接下来，处理振幅数据，以强调在幅度上升的语音信号部分中的有声帧的共振峰的峰值以及无条件地强调高频范围上的频谱包络。 通过基于强调幅度数据合成余弦波来产生浊音语音频谱。
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29.发明授权Method and apparatus for decoding and changing the pitch of an encoded speech signal 失效用于对编码语音信号进行解码和改变音调的方法和装置
公开(公告)号：US5873059A
公开(公告)日：1999-02-16
申请号：US736989
申请日：1996-10-25
CPC分类号： G10L13/033 , G10L21/01 , G10L19/087
摘要： A method and apparatus for reproducing speech signals at a controlled speed and for synthesizing speech includes a dividing unit that divides the input speech into time segments and an encoding unit that discriminates whether each of the speech segments is voiced or unvoiced. Based on the results of the discrimination, the encoding unit performs sinusoidal synthesis and encoding for voiced segments and vector quantization by closed-loop search for an optimum vector using an analysis-by-synthesis method for unvoiced segments in order to find encoded parameters. A period modification unit modifies the length of time associated with each signal segment and calculates a set of modified encoded parameters. In the speech synthesizing unit, encoded speech signal data is output from the encoding unit and pitch data and amplitude data specifying the spectral envelope are sent via a data conversion unit to a waveform synthesis unit, where the number of amplitude data points of the spectral envelope is changed without changing the shape of the spectral envelope, so that the pitch of the signal may be varied without changing its phoneme. A waveform synthesis unit synthesizes the speech waveform based on the converted spectral envelope data and pitch data.
摘要翻译： 用于以受控速度再现语音信号并用于合成语音的方法和装置包括将输入语音划分成时间段的分割单元和鉴别每个语音段是有声还是无声的编码单元。 基于鉴别的结果，编码单元通过使用用于清音段的合成分析方法对最佳向量进行闭环搜索，对浊音段和矢量量化进行正弦合成和编码，以便找到编码参数。 周期修改单元修改与每个信号段相关联的时间长度，并计算一组经修改的编码参数。 在语音合成单元中，编码语音信号数据从编码单元输出，音调数据和指定频谱包络的振幅数据经由数据转换单元发送到波形合成单元，其中频谱包络的振幅数据点的数量 在不改变频谱包络的形状的情况下改变，使得信号的音调可以改变而不改变其音素。 波形合成单元基于转换的频谱包络数据和音调数据来合成语音波形。
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30.发明授权Apparatus and method for encoding/decoding a speech signal using adaptively changing codebook vectors 失效使用自适应变化的码本矢量对语音信号进行编码/解码的装置和方法
公开(公告)号：US5828996A
公开(公告)日：1998-10-27
申请号：US736988
申请日：1996-10-25
摘要： An encoding apparatus in which an input speech signal is divided into blocks and encoded in units of blocks. The encoding apparatus includes an encoding unit for performing CELP encoding having a noise codebook memory containing having codebook vectors generated by clipping Gaussian noise and codebook vectors obtained by learning using the code vectors generated by clipping the Gaussian noise as initial values. The encoding apparatus enables optimum encoding for a variety of speech configurations.
摘要翻译： 一种编码装置，其中输入语音信号被分成块并以块为单位编码。 该编码装置包括编码单元，用于执行CELP编码，该编码单元具有噪声码本存储器，该噪声码本存储器包含通过使用通过限幅高斯噪声产生的代码矢量进行学习而获得的通过削波高斯噪声和码本矢量生成的码本矢量作为初始值。 编码装置能够对各种语音配置进行最佳编码。
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