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    • 1. 发明授权
    • Fixed sound source vector generation method and fixed sound source codebook
    • 固定声源矢量生成方法和固定声源码本
    • US07580834B2
    • 2009-08-25
    • US10505100
    • 2003-02-20
    • Hiroyuki EharaKazutoshi YasunagaKazunori ManoYusuke Hiwasaki
    • Hiroyuki EharaKazutoshi YasunagaKazunori ManoYusuke Hiwasaki
    • G10L19/00
    • G10L19/12
    • At the speech encoding end, upon generation of an fixed excitation vector, the shape of an excitation vector output from pulse excitation codebook 301 is identified in pulse excitation vector shape identifier 302, a dispersion vector used for excitation vectors of the shape is output from dispersion vector storage 304, and, in dispersion vector convolution processor 303, dispersion vector convolution processing of the excitation vector is performed. In particular, when a pulse excitation vector having a specific shape of high frequency of use is output from pulse excitation codebook 301, pulse excitation vector shape identifier 302 controls dispersion vector storage 304 in such a way that an additional dispersion vector prepared dedicated to the pulse excitation vector is output. By this means, it is possible to provide a technology that improves the quality of decoded speech and that decodes speech more natural and audible to the user.
    • 在语音编码结束时,在产生固定的激励矢量时,从脉冲激励码本301输出的激励矢量的形状在脉冲激励矢量形状识别符302中被识别,用于形状的激励矢量的色散矢量从色散 向量存储器304,并且在色散向量卷积处理器303中,执行激励矢量的色散向量卷积处理。 特别地,当从脉冲激励码本301输出具有高频用特定形状的脉冲激励矢量时,脉冲激励矢量形状识别器302以这样一种方式控制色散矢量存储器304,使得专用于脉冲的附加色散矢量 输出激励矢量。 通过这种方式,可以提供提高解码语音质量的技术,并且使用户更自然和可听地解码语音。
    • 5. 发明申请
    • Fixed sound source vector generation method and fixed sound source codebook
    • 固定声源矢量生成方法和固定声源码本
    • US20050228652A1
    • 2005-10-13
    • US10505100
    • 2003-02-20
    • Hiroyuki EharaKazutoshi YasunagaKazunori ManoYusuke Hiwasaki
    • Hiroyuki EharaKazutoshi YasunagaKazunori ManoYusuke Hiwasaki
    • G10L19/04G10L19/10
    • G10L19/12
    • At the speech encoding end, upon generation of an fixed excitation vector, the shape of an excitation vector output from pulse excitation codebook 301 is identified in pulse excitation vector shape identifier 302, a dispersion vector used for excitation vectors of the shape is output from dispersion vector storage 304, and, in dispersion vector convolution processor 303, dispersion vector convolution processing of the excitation vector is performed. In particular, when a pulse excitation vector having a specific shape of high frequency of use is output from pulse excitation codebook 301, pulse excitation vector shape identifier 302 controls dispersion vector storage 304 in such a way that an additional dispersion vector prepared dedicated to the pulse excitation vector is output. By this means, it is possible to provide a technology that improves the quality of decoded speech and that decodes speech more natural and audible to the user.
    • 在语音编码结束时,在产生固定的激励矢量时,从脉冲激励码本301输出的激励矢量的形状在脉冲激励矢量形状识别符302中被识别,用于形状的激励矢量的色散矢量从色散 向量存储器304,并且在色散向量卷积处理器303中,执行激励矢量的色散向量卷积处理。 特别地,当从脉冲激励码本301输出具有高频用特定形状的脉冲激励矢量时,脉冲激励矢量形状识别器302以这样一种方式控制色散矢量存储器304,使得专用于脉冲的附加色散矢量 输出激励矢量。 通过这种方式,可以提供提高解码语音质量的技术,并且使用户更自然和可听地解码语音。
    • 6. 发明授权
    • Audio coding and decoding methods and apparatuses and recording medium having recorded thereon programs for implementing them
    • 音频编码和解码方法和装置以及记录有用于实现它们的程序的记录介质
    • US06810381B1
    • 2004-10-26
    • US09568810
    • 2000-05-11
    • Shigeaki SasakiKazunori ManoShinji Hayashi
    • Shigeaki SasakiKazunori ManoShinji Hayashi
    • G10L1900
    • G10L19/18G10L19/06
    • In the CELP coding system a low-order synthesis filter and a cascade-connected synthesis filter formed by a cascade connection of low- and high-order synthesis filters are provided, a synthesized acoustic signal is estimated in a mode decision part for an input acoustic signal, and the estimated synthesized acoustic signal is subjected to inverse filtering by an inverse filter corresponding to the low-order synthesis filter and an inverse filter corresponding to the cascade-connected synthesis filter to obtain residual signals. That one of the synthesis filters which corresponds to the residual signal of smaller power is selected by a switch, and a codebook is searched for indices which will minimize the error between the output synthesized acoustic signal by the selected synthesis filter and the input acoustic signal.
    • 在CELP编码系统中,提供了通过低阶和高阶合成滤波器的级联连接形成的低阶合成滤波器和级联连接的合成滤波器,在用于输入声学的模式判定部分中估计合成的声信号 信号,并且通过对应于低阶合成滤波器的逆滤波器和对应于级联连接的合成滤波器的逆滤波器对估计的合成声信号进行逆滤波,以获得残留信号。 通过开关选择对应于较小功率的残余信号的合成滤波器中的一个,并且搜索码本将使所选合成滤波器的输出合成声信号与输入声信号之间的误差最小化的索引。
    • 7. 发明授权
    • Speech coding by code-edited linear prediction
    • 通过编码线性预测的语音编码
    • US5787391A
    • 1998-07-28
    • US658303
    • 1996-06-05
    • Takehiro MoriyaAkitoshi KataokaKazunori ManoSatoshi MikiHitoshi OmuroShinji Hayashi
    • Takehiro MoriyaAkitoshi KataokaKazunori ManoSatoshi MikiHitoshi OmuroShinji Hayashi
    • G10L19/005G10L19/06G10L19/07G10L19/08G10L19/083G10L19/12G10L19/135G10L3/02
    • G10L19/08G10L19/005G10L19/06G10L19/07G10L19/083G10L19/12G10L19/135
    • In a speech coding method of the present invention, initially, a plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated. Then, the prediction coefficients are quantized, and the quantized prediction coefficients are set in a synthesis filter. Moreover, a pitch period vector is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and the selected pitch period vector is multiplied by a first gain which is obtained, at the same time, with a second gain. In addition, a noise waveform vector is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and is multiplied by a predicted gain and the second gain. Then, the speech vector is synthesized by exciting the synthesis filter with the pitch period vector multiplied by the first gain, and with the noise waveform vector multiplied by the predicted gain and the second gain. Consequently, speech data comprising a plurality of samples are coded as a unit of a frame operation. Furthermore, the predicted gain multiplied by the noise waveform vector which is selected in a subsequent frame operation, is predicted based on the current noise waveform vector which is multiplied by the predicted gain and the second gain at the current frame operation, and also the previous waveform vector which is multiplied by the predicted gain and the second gain in the previous frame operation.
    • 在本发明的语音编码方法中,首先,通过线性预测分析来分析多个语音数据样本,从而计算出预测系数。 然后,量化预测系数,并将量化的预测系数设置在合成滤波器中。 此外,从存储多个音调周期矢量的自适应码本中选择音调周期矢量,并且将所选择的音调周期矢量乘以与第二增益同时获得的第一增益。 此外,从存储多个噪声波形向量的随机码本中选择噪声波形向量,并将其乘以预测的增益和第二增益。 然后,通过利用乘以第一增益的音调周期矢量并且噪声波形向量乘以预测增益和第二增益来激励合成滤波器来合成语音向量。 因此,包括多个样本的语音数据被编码为帧操作的单位。 此外,基于在当前帧操作中乘以预测增益和第二增益的当前噪声波形向量来预测在后续帧操作中选择的噪声波形向量的预测增益,以及前一帧 在前一帧操作中乘以预测增益和第二增益的波形向量。