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    • 1. 发明授权
    • Audio encoding apparatus and spectrum modifying method
    • 音频编码装置和频谱修改方法
    • US08296134B2
    • 2012-10-23
    • US11914296
    • 2006-05-11
    • Chun Woei TeoSua Hong NeoKoji YoshidaMichiyo Goto
    • Chun Woei TeoSua Hong NeoKoji YoshidaMichiyo Goto
    • G10L19/14
    • G10L19/0204G10L19/008G10L19/09
    • A spectrum modifying method and the like wherein the efficiencies of the signal estimation and prediction can be improved and the spectrum can be more efficiently encoded. According to this method, the pitch period is calculated from an original signal, which serves as a reference signal, and then a basic pitch frequency (f0) is calculated. Thereafter, the spectrum of a target signal, which is a target of spectrum modification, is divided into a plurality of partitions. It is specified here that the width of each partition be the basic pitch frequency. Then, the spectra of bands are interleaved such that a plurality of peaks having similar amplitudes are unified into a group. The basic pitch frequency is used as an interleave pitch.
    • 频谱修正方法等可以改善信号估计和预测的效率,并且可以更有效地编码频谱。 根据这种方法,从作为参考信号的原始信号计算音调周期,然后计算基本音调频率(f0)。 此后,将作为频谱修改对象的目标信号的频谱划分成多个分区。 这里指定每个分区的宽度为基本音调频率。 然后,频带的频谱被交织,使得具有相似幅度的多个峰值被统一成一组。 基本音调频率用作交织音调。
    • 2. 发明授权
    • Stereo signal generating apparatus and stereo signal generating method
    • 立体声信号发生装置和立体声信号产生方法
    • US08019087B2
    • 2011-09-13
    • US11573760
    • 2005-08-29
    • Michiyo GotoChun Woei TeoSua Hong NeoKoji Yoshida
    • Michiyo GotoChun Woei TeoSua Hong NeoKoji Yoshida
    • H04R5/00G10L19/00G10L13/00
    • G10L19/008
    • A stereo signal generating apparatus capable of obtaining stereo signals that exhibit a low bit rate and an excellent reproducibility. In this stereo signal generating apparatus (90), an FT part (901) converts a monaural signal (M′t) of time domain to a monaural signal (M′) of frequency domain. A power spectrum calculating part (902) determines a power spectrum (PM′). A scaling ratio calculating part (904a) determines a scaling ratio (SL) for a left channel, while a scaling ratio calculating part (904b) determines a scaling ratio (SR) for a right channel. A multiplying part (905a) multiplies the monaural signal (M′) of frequency domain by the scaling ratio (SL) to produce a left channel signal (L″) of a stereo signal, while a multiplying part (905b) multiplies the monaural signal (M′) of frequency domain by the scaling ratio (SR) to produce a right channel signal (R″) of the stereo signal.
    • 一种立体声信号发生装置,其能够获得表现出低比特率和优异的再现性的立体声信号。 在该立体声信号生成装置(90)中,FT部(901)将时域的单声道信号(M't)变换为频域的单声道信号(M')。 功率谱计算部(902)决定功率谱(PM')。 缩放比例计算部分(904a)确定左声道的缩放率(SL),而缩放比例计算部分(904b)确定右声道的缩放比率(SR)。 乘法部分(905a)将频域的单声道信号(M')乘以缩放比率(SL)以产生立体声信号的左声道信号(L“),而乘法部分(905b)将单声道信号 (M')通过缩放比(SR)产生立体声信号的右声道信号(R“)。
    • 3. 发明申请
    • STEREO AUDIO ENCODING DEVICE, STEREO AUDIO DECODING DEVICE, AND METHOD THEREOF
    • 立体声音频编码装置,立体声音频解码装置及其方法
    • US20100010811A1
    • 2010-01-14
    • US12376025
    • 2007-08-02
    • Jiong ZhouSua Hong NeoKoji YoshidaMichiyo Goto
    • Jiong ZhouSua Hong NeoKoji YoshidaMichiyo Goto
    • G10L19/00
    • G10L19/008G10L19/04
    • Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit (103) performs LPC analysis on an L channel signal and an R channel signal so as to obtain an L channel LPC coefficient and an R channel LPC coefficient. An LPC coefficient adaptive filter (105) obtains an LPC coefficient adaptive filter parameter to minimize the mean square error between the L channel LPC coefficient and the R channel LPC coefficient. An LPC coefficient reconfiguration unit (106) reconfigures the R channel LPC coefficient by using the L channel LPC coefficient and the LPC coefficient adaptive filter parameter. A route calculation unit (107) calculates a polynomial route indicating the safety of the R channel reconfigured LPC coefficient. A selection unit (108) selects and outputs the LPC coefficient adaptive filter parameter or the R channel LPC coefficient according to the safety of the R channel reconfigured LPC coefficient.
    • 公开了能够降低比特率的立体声音频编码装置。 在该装置中,立体声音频编码单元(103)对L声道信号和R声道信号执行LPC分析,以获得L声道LPC系数和R声道LPC系数。 LPC系数自适应滤波器(105)获得LPC系数自适应滤波器参数以最小化L信道LPC系数和R信道LPC系数之间的均方误差。 LPC系数重构单元(106)通过使用L信道LPC系数和LPC系数自适应滤波器参数来重新配置R信道LPC系数。 路线计算单元(107)计算表示R信道重新配置的LPC系数的安全性的多项式路由。 选择单元(108)根据R信道重新配置的LPC系数的安全性来选择并输出LPC系数自适应滤波器参数或R信道LPC系数。
    • 4. 发明申请
    • STEREO ENCODING DEVICE, AND STEREO SIGNAL PREDICTING METHOD
    • 立体声编码装置和立体声信号预测方法
    • US20090119111A1
    • 2009-05-07
    • US12091793
    • 2006-10-30
    • Michiyo GotoKoji YoshidaHiroyuki Ehara
    • Michiyo GotoKoji YoshidaHiroyuki Ehara
    • G10L19/04
    • G10L19/008G10L25/12
    • A prediction performance between the individual channels of a stereo signal is improved to improve the sound quality of a decoded signal. An LPF (101-1) interrupts the high-range component of an S1, and outputs an S1′ (a low-range component). An LPF (101-2) interrupts the high-range component of an S2, and outputs an S2′ (a low-range component). A prediction unit (102) predicts the S2′ from the S1′, and outputs a prediction parameter composed of a delay time difference (t) and an amplitude ratio (g). A first channel encoding unit (103) encodes the S1. A prediction parameter encoding unit (104) encodes the prediction parameter. The encoded parameters of the encoded parameter of the S1 and the prediction parameter are finally outputted.
    • 提高了立体声信号的各声道之间的预测性能,以提高解码信号的声音质量。 LPF(101-1)中断S1的高范围分量,并输出S1'(低范围分量)。 LPF(101-2)中断S2的高范围分量,并输出S2'(低范围分量)。 预测单元(102)从S1'预测S2',并输出由延迟时间差(t)和振幅比(g)组成的预测参数。 第一信道编码单元(103)对S1进行编码。 预测参数编码单元(104)对预测参数进行编码。 最后输出S1的编码参数的编码参数和预测参数。
    • 5. 发明申请
    • SCALABLE ENCODING DEVICE AND SCALABLE ENCODING METHOD
    • 可扩展编码设备和可扩展编码方法
    • US20090041255A1
    • 2009-02-12
    • US11815028
    • 2006-01-30
    • Michiyo GotoKoji Yoshida
    • Michiyo GotoKoji Yoshida
    • H04R5/00
    • G10L19/24G10L19/008G10L19/12
    • There is disclosed a scalable encoding device capable of preventing sound quality deterioration of a decoded signal, reducing the encoding rate, and reducing the circuit size. The scalable encoding device includes: a first layer encoder (100) for generating a monaural signal by using a plurality of channel signals (L channel signal and R channel signal) constituting a stereo signal and encoding the monaural signal to generate a sound source parameter; and a second layer encoder (150) for generating a first conversion signal by using the channel signal and the monaural signal, generating a synthesis signal by using the sound source parameter and the first conversion signal, and generating a second conversion coefficient index by using the synthesis signal and the first conversion signal.
    • 公开了一种能够防止解码信号的声音劣化,降低编码率,降低电路尺寸的可伸缩编码装置。 可伸缩编码装置包括:第一层编码器(100),用于通过使用构成立体声信号的多个声道信号(L声道信号和R声道信号)产生单声道信号,并对单声道信号进行编码以产生声源参数; 以及第二层编码器(150),用于通过使用所述通道信号和所述单声道信号产生第一转换信号,通过使用所述声源参数和所述第一转换信号产生合成信号,并且通过使用所述第一转换系数索引 合成信号和第一转换信号。
    • 6. 发明申请
    • Audio Encoding Device and Audio Encoding Method
    • 音频编码设备和音频编码方法
    • US20080091419A1
    • 2008-04-17
    • US11722821
    • 2005-12-26
    • Koji YoshidaMichiyo Goto
    • Koji YoshidaMichiyo Goto
    • G10L13/00
    • G10L19/008
    • There is provided an audio encoding device capable of generating an appropriate monaural signal from a stereo signal while suppressing the lowering of encoding efficiency of the monaural signal. In a monaural signal generation unit (101) of this device, an inter-channel prediction/analysis unit (201) obtains a prediction parameter based on a delay difference and an amplitude ratio between a first channel audio signal and a second channel audio signal; an intermediate prediction parameter generation unit (202) obtains an intermediate parameter of the prediction parameter (called intermediate prediction parameter) so that the monaural signal generated finally is an intermediate signal of the first channel audio signal and the second channel audio signal; and a monaural signal calculation unit (203) calculates a monaural signal by using the intermediate prediction parameter.
    • 提供一种音频编码装置,其能够在抑制单声道信号的编码效率降低的同时,从立体声信号生成适当的单声道信号。 在该设备的单声道信号生成单元(101)中,信道间预测/分析单元(201)基于第一声道音频信号和第二声道音频信号之间的延迟差和幅度比来获得预测参数; 中间预测参数生成单元(202)获取预测参数(称为中间预测参数)的中间参数,使得最终生成的单声道信号是第一声道音频信号和第二声道音频信号的中间信号; 和单声道信号计算单元(203)通过使用中间预测参数来计算单声道信号。
    • 7. 发明申请
    • Sound Coding Device and Sound Coding Method
    • 声音编码装置和声音编码方法
    • US20080010072A1
    • 2008-01-10
    • US11722737
    • 2005-12-26
    • Koji YoshidaMichiyo Goto
    • Koji YoshidaMichiyo Goto
    • G10L21/00
    • G10L19/008G10L19/24
    • A sound coding device having a monaural/stereo scalable structure and capable of efficiently coding stereo sound. even when the correlation between the channel signals of a stereo signal is small. In a core layer coding block (110) of this device, a monaural signal generating section (111) generates a monaural signal from first and second-channel sound signal, a monaural signal coding section (112) codes the monaural signal, and a monaural signal decoding section (113) greatest a monaural decoded signal from monaural signal coded data and outputs it to an expansion layer coding block (120). In the expansion layer coding block (120), a first-channel prediction signal synthesizing section (122) synthesizes a first-channel prediction signal from the monaural decoded signal and a first-channel prediction filter digitizing parameter and a second-channel prediction signal synthesizing section (126) synthesizes a second-channel prediction signal from the monaural decoded signal and second-channel prediction filter digitizing parameter.
    • 一种声音编码装置,具有单声道/立体声可缩放结构,并能够有效地编码立体声。 即使立体声信号的信道信号之间的相关性小。 在该装置的核心层编码块(110)中,单声道信号生成部(111)从第一和第二声道声音信号生成单声道信号,单声道信号编码部(112)对单声道信号进行编码, 信号解码部分(113)对来自单声信号编码数据的单声道解码信号最大并将其输出到扩展层编码块(120)。 在扩展层编码块(120)中,第一信道预测信号合成部(122)从单声道解码信号和第一信道预测滤波数字化参数以及合成的第二信道预测信号合成第一信道预测信号 部分(126)从单声道解码信号和第二声道预测滤波器数字化参数合成第二声道预测信号。
    • 8. 发明授权
    • Scalable encoding device and scalable encoding method
    • 可扩展编码设备和可扩展编码方法
    • US08036390B2
    • 2011-10-11
    • US11815028
    • 2006-01-30
    • Michiyo GotoKoji Yoshida
    • Michiyo GotoKoji Yoshida
    • H04R5/00
    • G10L19/24G10L19/008G10L19/12
    • A scalable encoding device prevents sound quality deterioration of a decoded signal, reduces the encoding rate, and reduces the circuit size. The scalable encoding device includes a first layer encoder for generating a monaural signal by using a plurality of channel signals (L channel signal and R channel signal) constituting a stereo signal and encoding the monaural signal to generate a sound source parameter. The scalable encoding device also includes a second layer encoder for generating a first conversion signal by using the channel signal and the monaural signal, generating a synthesis signal by using the sound source parameter and the first conversion signal, and generating a second conversion coefficient index by using the synthesis signal and the first conversion signal.
    • 可扩展编码装置防止解码信号的声音质量恶化,降低编码速率,并减小电路尺寸。 可扩展编码装置包括:第一层编码器,用于通过使用构成立体声信号的多个声道信号(L声道信号和R声道信号)产生单声道信号,并对单声道信号进行编码以产生声源参数。 可扩展编码装置还包括第二层编码器,用于通过使用信道信号和单声道信号产生第一转换信号,通过使用声源参数和第一转换信号产生合成信号,并通过 使用合成信号和第一转换信号。
    • 9. 发明申请
    • SPEECH ENCODING APPARATUS AND SPEECH ENCODING METHOD
    • 语音编码装置和语音编码方法
    • US20100153099A1
    • 2010-06-17
    • US12088318
    • 2006-09-29
    • Michiyo GotoKoji Yoshida
    • Michiyo GotoKoji Yoshida
    • G10L19/00
    • G10L19/02G10L19/12
    • A speech coder and so forth for preventing deterioration of the quality of a reproduced speech signal while reducing the coding rate. In a speech signal modifying section (101) of the coder, a masking threshold calculating section (114) calculates a masking threshold M(f)) of the spectrum S(f) of an input speech signal, an ACB sound source model spectrum calculating section (117) calculates an adaptive codebook sound source model spectrum SACB(f), an input spectrum shape modifying section (112) refers to both values of the masking threshold M(f) and the adaptive code book sound source model spectrum S′ACB(f) having an LPC spectral envelope and carries out a preprocessing of the spectrum S(f) so that the shape of the spectrum S(f) is modified to match a CELP coding section (102) of the succeeding stage. The CELP coding section (102) carries out CELP coding of the preprocessed speech signal and outputs a coded parameter.
    • 一种用于防止再现的语音信号的质量恶化同时降低编码率的语音编码器等。 在编码器的语音信号修改部分(101)中,掩蔽阈值计算部分(114)计算输入语音信号的频谱S(f)的掩蔽阈值M(f)),ACB声源模型频谱计算 部分(117)计算自适应码本声源模型谱SACB(f),输入谱形状修正部分(112)参考掩蔽阈值M(f)和自适应码本声源模型谱S'ACB (f)具有LPC频谱包络并执行频谱S(f)的预处理,使得频谱S(f)的形状被修改以匹配后一级的CELP编码部分(102)。 CELP编码部(102)对预处理语音信号进行CELP编码,并输出编码参数。