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    • 91. 发明申请
    • APPARATUS AND METHOD FOR PROCESSING SIGNAL, RECORDING MEDIUM, AND PROGRAM
    • 用于处理信号,记录媒体和程序的装置和方法
    • US20080082343A1
    • 2008-04-03
    • US11844784
    • 2007-08-24
    • Yuuji MAEDA
    • Yuuji MAEDA
    • G10L19/04
    • G10L19/005G10L19/08
    • A signal processing apparatus includes a decoding unit, an analyzing unit, a synthesizing unit, and a selecting unit. The decoding unit decodes an input encoded audio signal and outputs a playback audio signal. When loss of the encoded audio signal occurs, the analyzing unit analyzes the playback audio signal output before the loss occurs and generates a linear predictive residual signal. The synthesizing unit synthesizes a synthesized audio signal on the basis of the linear predictive residual signal. The selecting unit selects one of the synthesized audio signal and the playback audio signal and outputs the selected audio signal as a continuous output audio signal.
    • 信号处理装置包括解码单元,分析单元,合成单元和选择单元。 解码单元解码输入的编码音频信号并输出​​回放音频信号。 当发生编码音频信号的丢失时,分析单元分析在丢失发生之前输出的重放音频信号,并产生线性预测残差信号。 合成单元基于线性预测残差信号合成合成音频信号。 选择单元选择合成音频信号和回放音频信号中的一个,并输出所选择的音频信号作为连续输出音频信号。
    • 93. 发明授权
    • Speech coding apparatus including enhancement layer performing long term prediction
    • 包括执行长期预测的增强层的语音编码装置
    • US07299174B2
    • 2007-11-20
    • US10554619
    • 2004-04-30
    • Kaoru SatoToshiyuki Morii
    • Kaoru SatoToshiyuki Morii
    • G10L19/04G10L19/12
    • G10L19/24
    • To implement scalable coding, a base layer coding section encodes an input signal to obtain base layer coded information, which is decoded by a base layer decoding section to obtain a base layer decoded signal and long term prediction information (pitch lag). An adding section inverts the polarity of the base layer decoded signal to add to the input signal, and obtains a residual signal. An enhancement layer coding section encodes a long term prediction coefficient calculated using the long term prediction information and the residual signal to obtain enhancement layer coded information. Also using the long term prediction information, an enhancement layer decoding section decodes the enhancement layer coded information to obtain an enhancement layer decoded signal. An adding section adds the base layer decoded signal and enhancement layer decoded signal to obtain a speech/sound signal.
    • 为了实现可分级编码,基层编码部分对输入信号进行编码以获得由基本层解码部分解码以获得基本层解码信号和长期预测信息(音调滞后)的基本层编码信息。 加法部分反转基本层解码信号的极性以加到输入信号上,并获得残留信号。 增强层编码部分编码使用长期预测信息和残差信号计算的长期预测系数,以获得增强层编码信息。 还使用长期预测信息,增强层解码部分对增强层编码信息进行解码以获得增强层解码信号。 加法部分添加基本层解码信号和增强层解码信号以获得语音/声音信号。
    • 96. 发明申请
    • Method and apparatus for modifying an encoded signal
    • 修改编码信号的方法和装置
    • US20060217971A1
    • 2006-09-28
    • US11159845
    • 2005-06-22
    • Rafid SukkarRichard YouncePeng Zhang
    • Rafid SukkarRichard YouncePeng Zhang
    • G10L19/04
    • G10L19/173G10L21/02
    • Voice Quality Enhancement is performed directly in a coded domain. At least one parameter of a first encoded signal is modified, which results in a corresponding at least one modified parameter. The at least one parameter of the first encoded signal is replaced with the at least one modified parameter, which results in a second encoded signal which, in a decoded state, approximates a target signal that is a function of at least the first encoded signal in at least a partially decoded state. Thus, the signal does not have to go through intermediate decoder/re-encoder(s), which can degrade overall speech quality. Computational resources required for a complete re-encoding are not needed. Overall delay of the system is minimized. The CD-VQE system can be used in any network in which signals are communicated in a coded domain, such as a Third Generation (3G) wireless network.
    • 语音质量增强直接在编码域中执行。 修改第一编码信号的至少一个参数,这导致对应的至少一个修改参数。 第一编码信号的至少一个参数被替换为至少一个修改的参数,其导致第二编码信号,其在解码状态下近似于至少是第一编码信号的函数的目标信号 至少部分解码状态。 因此,信号不必经过中间解码器/重新编码器,这可能降低整体语音质量。 不需要完全重新编码所需的计算资源。 系统的整体延迟最小化。 CD-VQE系统可以用于信号在诸如第三代(3G)无线网络的编码域中传送的任何网络中。
    • 98. 发明授权
    • Method and device for coding speech in analysis-by-synthesis speech coders
    • 用于分析综合语音编码器中的语音编码的方法和装置
    • US07089180B2
    • 2006-08-08
    • US10167287
    • 2002-06-10
    • Ari P. Heikkinen
    • Ari P. Heikkinen
    • G10L19/04
    • G10L19/10
    • The present invention discloses a method of improving the coded speech quality in low bit rate analysis-by-synthesis (AbS) speech coders. This is accomplished by relaxing the waveform matching constraints for non-stationary plosive speech segments of speech signals by suitably shifting pulse locations of the coded excitation signal. The shifting results in the coded signal having phase information that does not exactly match original signal in places where it is perceptually insignificant to the listener. Furthermore, a technique for adaptive phase dispersion is introduced to the coded excitation signal to efficiently preserve important signal characteristics such as the energy spread of the original signal.
    • 本发明公开了一种在低比特率综合(AbS)语音编码器中提高编码语音质量的方法。 这是通过适当地移动编码的激励信号的脉冲位置来放宽语音信号的非静止爆音语音段的波形匹配约束来实现的。 移位导致编码信号具有在听觉上对收听者无意义的地方与原始信号不完全匹配的相位信息。 此外,用于自适应相位色散的技术被引入编码的激励信号,以有效地保持诸如原始信号的能量扩展的重要信号特性。
    • 99. 发明授权
    • Audio coder
    • 音频编码器
    • US07072830B2
    • 2006-07-04
    • US11185302
    • 2005-07-20
    • Hitoshi SasakiYasuji Ota
    • Hitoshi SasakiYasuji Ota
    • G10L19/04
    • G10L19/032
    • An audio coder that improves audio quality by reducing a quantization error. When a code corresponding to a sampled value of an audio signal is determined, a candidate code storage section stores all combinations of candidate codes in a neighborhood interval of the sampled value. A local decoder generates reproduced signals by decoding the codes stored in the candidate code storage section. An error evaluation section calculates, for each candidate code, a sum of squares of differentials between input sampled values and reproduced signals, detects a combination of candidate codes by which a smallest sum is obtained, that is to say, which minimizes a quantization error, and outputs a code included in the detected combination of candidate codes.
    • 一种通过减少量化误差来提高音频质量的音频编码器。 当确定与音频信号的采样值相对应的代码时,候选代码存储部分将候选代码的所有组合存储在采样值的邻域间隔中。 本地解码器通过解码存储在候选代码存储部分中的代码来产生再现信号。 误差评估部分针对每个候选代码计算输入采样值和再现信号之间的差分的平方和,检测获得最小和的候选代码的组合,即将量化误差最小化的组合, 并且输出检测到的候选代码的组合中包括的代码。
    • 100. 发明授权
    • Computer-implemented method and apparatus for audio data hiding
    • 用于音频数据隐藏的计算机实现的方法和装置
    • US07058570B1
    • 2006-06-06
    • US09499525
    • 2000-02-10
    • Hong Heather YuXin Li
    • Hong Heather YuXin Li
    • G10L19/04
    • G10L19/018G10L25/24
    • A computer-implemented method and apparatus for embedding hidden data in an audio signal. An audio signal is received in a base domain and then transformed into a non-base domain, such as cepstrum domain or LP residue domain. The statistical mean manipulation is employed on selected transform coefficients to embed hidden data. The introduced distortion is controlled by psychoacoustic model to ensure the imperceptibility of the embedded hidden data. Scrambling techniques can be plugged in to further increase the security of the data hiding system. The present new audio data hiding scheme provides transparent audio quality, sufficient embedding capacity, and high survivability over a wide range of common signal processing attacks.
    • 一种用于将隐藏数据嵌入音频信号中的计算机实现的方法和装置。 在基本域中接收音频信号,然后将其转换成非碱基域,如倒谱域或LP残基域。 在选择的变换系数上采用统计平均值操纵来嵌入隐藏数据。 引入的失真由心理声学模型控制,以确保嵌入式隐藏数据的不可察觉性。 可以插入加扰技术,进一步提高数据隐藏系统的安全性。 现有的新的音频数据隐藏方案提供透明的音频质量,足够的嵌入容量,以及广泛的常见信号处理攻击的高生存能力。