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    • 2. 发明授权
    • Quantizer, encoder, and the methods thereof
    • 量化器,编码器及其方法
    • US08473288B2
    • 2013-06-25
    • US12990697
    • 2009-06-18
    • Toshiyuki MoriiHiroyuki EharaKoji Yoshida
    • Toshiyuki MoriiHiroyuki EharaKoji Yoshida
    • G10L19/00
    • G10L19/008G10L25/27
    • Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer includes a power correlation calculator which calculates the power of the left channel signal, the power of the right channel signal, and the correlation between the left channel signal and the right channel signal; an intermediate value calculator which calculates the intermediate value which is the difference between left channel signal the power and the right channel signal power; a codebook which holds a plurality of sets of the coefficients related to the transform coefficients of the principal component analysis transform and the code; and a quantizer which calculates the sum of the first multiplication result obtained by multiplying the coefficient by the correlation value and the second multiplication result obtained by multiplying the coefficient by the intermediate value as the cost function E, selects the coefficients where the cost function E becomes the maximum, and fetches the code related to the selected coefficients as the quantized code.
    • 公开了一种量化器,编码器及其方法,其中当将主分量分析变换应用于代码立体声时,当与主成分分析变换的变换系数相关的值被量化时,计算负荷减小。 量化器包括功率相关计算器,其计算左声道信号的功率,右声道信号的功率,以及左声道信号和右声道信号之间的相关性; 中间值计算器,其计算作为左声道信号的功率和右声道信号功率之间的差的中间值; 码本,其保存与主成分分析变换和代码的变换系数相关的多个系数集合; 以及量化器,其计算通过将系数乘以相关值而获得的第一相乘结果与通过将系数乘以中间值而获得的第二乘法结果作为成本函数E的和,选择成本函数E变为 最大值,并且将与所选择的系数相关的代码作为量化代码获取。
    • 3. 发明授权
    • Speech encoding apparatus and speech encoding method
    • 语音编码装置和语音编码方法
    • US08239191B2
    • 2012-08-07
    • US12440661
    • 2007-09-14
    • Hiroyuki EharaToshiyuki MoriiKoji Yoshida
    • Hiroyuki EharaToshiyuki MoriiKoji Yoshida
    • G10L19/00G10L21/02
    • G10L19/265G10L19/08
    • Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.
    • 公开了能够在不改变共振峰重量的情况下调整量化噪声的频谱倾斜度的音频编码装置。 该装置包括:HPF(131),其从输入音频信号中提取频率区域的高频分量; 高频能量计算单元,其计算帧单位中的高频分量的能级; LPF(133),其从所述输入音频信号中提取所述频率区域的低频分量; 低能量计算单元(134),其计算帧单位中的低频分量的能级; 倾斜校正系数计算单元(141)将从加法器(140)输入的低频分量的SNR与高频成分的SNR之差乘以常数,并将偏置分量加到乘积上,以计算 倾斜校正系数α3。 倾斜校正系数用于调整量化噪声的频谱倾斜度。
    • 5. 发明申请
    • SPEECH ENCODING APPARATUS AND SPEECH ENCODING METHOD
    • 语音编码装置和语音编码方法
    • US20090265167A1
    • 2009-10-22
    • US12440661
    • 2007-09-14
    • Hiroyuki EharaToshiyuki MoriiKoji Yoshida
    • Hiroyuki EharaToshiyuki MoriiKoji Yoshida
    • G10L19/08G10L19/14
    • G10L19/265G10L19/08
    • Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.
    • 公开了能够在不改变共振峰重量的情况下调整量化噪声的频谱倾斜度的音频编码装置。 该装置包括:HPF(131),其从输入音频信号中提取频率区域的高频分量; 高频能量计算单元,其计算帧单位中的高频分量的能级; LPF(133),其从所述输入音频信号中提取所述频率区域的低频分量; 低能量计算单元(134),其计算帧单位中的低频分量的能级; 倾斜校正系数计算单元(141)将从加法器(140)输入的低频分量的SNR与高频成分的SNR之差乘以常数,并将偏置分量加到乘积上,以计算 倾斜校正系数α3。 倾斜校正系数用于调整量化噪声的频谱倾斜度。
    • 6. 发明申请
    • QUANTIZER, ENCODER, AND THE METHODS THEREOF
    • 量子,编码器及其方法
    • US20110125495A1
    • 2011-05-26
    • US12990697
    • 2009-06-18
    • Toshiyuki MoriiHiroyuki EharaKoji Yoshida
    • Toshiyuki MoriiHiroyuki EharaKoji Yoshida
    • G10L19/00
    • G10L19/008G10L25/27
    • Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer (110) is comprised of a power correlation calculation unit (111) which calculates the power (C11) of the left channel signal, the power (C22) of the right channel signal, and the correlation (C12) between the left channel signal and the right channel signal; an intermediate value calculation unit (112) which calculates the intermediate value (C1122) which is the difference between the power (C11) and the power (C22); a codebook (113) which holds a plurality of sets of the coefficients ?1,n,?2,n related to the transform coefficients of the principal component analysis transform and the code; and a quantizer (114) which calculates the sum of the first multiplication result obtained by multiplying the coefficient ?1,n by the correlation value C12 and the second multiplication result obtained by multiplying the coefficient ?1,n by the intermediate value C1122 as the cost function E, selects the coefficients ?1,n,?2,n where the cost function E becomes the maximum, and fetches the code related to the selected coefficients ?1,n,?2,n as the quantized code.
    • 公开了一种量化器,编码器及其方法,其中当将主分量分析变换应用于代码立体声时,当与主成分分析变换的变换系数相关的值被量化时,计算负荷减小。 量化器(110)包括功率相关计算单元(111),其计算左声道信号的功率(C11),右声道信号的功率(C22)和左声道信号之间的相关性(C12) 信号和右声道信号; 计算作为功率(C11)和功率(C22)之间的差的中间值(C1122)的中间值计算单元(112)。 保存与主成分分析变换和代码的变换系数有关的多组系数α1,n,α2,n的码本(113); 以及量化器(114),其计算通过将系数α1,n乘以相关值C12获得的第一相乘结果与通过将系数α1,n乘以中间值C1122而获得的第二乘法结果的和作为 成本函数E选择成本函数E变为最大的系数α1,n,β2,n,并且取出与所选择的系数α1,n,φ2,n相关的代码作为量化代码。
    • 7. 发明授权
    • CELP speech decoder modifying an input vector with a fixed waveform to transform a waveform of the input vector
    • CELP语音解码器用固定波形修改输入向量,以变换输入矢量的波形
    • US08036887B2
    • 2011-10-11
    • US12781049
    • 2010-05-17
    • Kazutoshi YasunagaToshiyuki MoriiHiroyuki Ehara
    • Kazutoshi YasunagaToshiyuki MoriiHiroyuki Ehara
    • G10L19/12
    • G10L19/135G10L19/12G10L2019/0007G10L2019/0013
    • A CELP speech decoder includes an adaptive codebook that generates an adaptive code vector and a random codebook that generates a random code vector. The random codebook includes an input vector provider that provides an input vector including at least one pulse, each pulse having a position and a polarity, a fixed waveform storage that stores at least one fixed waveform, and a selector that selects at least one of a first process and a second process based on a value of an adaptive codebook gain. The random codebook further includes a convolution section that generates the random code vector by convoluting the at least one fixed waveform with the input vector when the first process is selected. A synthesis filter outputs synthesized speech by performing linear prediction coefficient synthesis on a signal based on the adaptive code vector and the random code vector.
    • CELP语音解码器包括产生自适应码矢量的自适应码本和产生随机码矢量的随机码本。 所述随机码本包括输入向量提供者,所述输入向量提供器提供包括至少一个脉冲,每个具有位置和极性的脉冲,存储至少一个固定波形的固定波形存储器的输入向量,以及选择器, 第一处理和基于自适应码本增益的值的第二处理。 所述随机码本还包括卷积部分,其通过在选择所述第一处理时将所述至少一个固定波形与所述输入矢量进行卷积来生成所述随机码矢量。 合成滤波器通过对基于自适应码矢量和随机码矢量的信号执行线性预测系数合成来输出合成语音。