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    • 2. 发明授权
    • Adaptive combining of multi-mode coding for voiced speech and noise-like signals
    • 用于有声语音和类噪声信号的多模式编码的自适应组合
    • US06192335B1
    • 2001-02-20
    • US09144961
    • 1998-09-01
    • Erik EkuddenRoar Hagen
    • Erik EkuddenRoar Hagen
    • G10L1104
    • G10L19/083G10L2019/0003G10L2025/935
    • In producing from an original speech signal a plurality of parameters from which an approximation of the original speech signal can be reconstructed, a coded signal of the original speech signal is generated. At least one of the parameters is determined using first and second differences between the original speech signal and the coded signal. The first difference is a difference between a waveform associated with the original speech signal and a waveform associated with the coded signal, and the second difference is a difference between an energy parameter derived from the original speech signal and a corresponding energy parameter associated with the coded signal.
    • 在从原始语音信号生成可以重构原始语音信号的近似的多个参数时,生成原始语音信号的编码信号。 使用原始语音信号和编码信号之间的第一和第二差异来确定参数中的至少一个。 第一个区别是与原始语音信号相关联的波形与与编码信号相关联的波形之间的差异,第二个差异是从原始语音信号导出的能量参数与与编码信号相关联的相应能量参数之间的差异 信号。
    • 3. 发明授权
    • High resolution post processing method for a speech decoder
    • 用于语音解码器的高分辨率后处理方法
    • US6138093A
    • 2000-10-24
    • US32942
    • 1998-03-02
    • Erik EkuddenRoar HagenBastiaan Kleijn
    • Erik EkuddenRoar HagenBastiaan Kleijn
    • G10L19/26G10L21/0232G10L25/27H03M7/30H04B14/00G10L19/02
    • G10L19/26G10L21/0232G10L25/27
    • A post-processing method for a speech decoder which outputs a decoded speech signal in the time domain provides high frequency resolution based on a frequency spectrum having non-harmonic and noise deficiencies. This is obtained by transforming the decoded time domain signal to a frequency domain signal by using a high frequency resolution transform (FFT). Then an analysis of the energy distribution of the frequency domain signal is made throughout its frequency area (4 kHz) to find the disturbing frequency components and to prioritize such frequency components which are situated in the higher part of the frequency spectrum. Next, the suppression degree for the disturbing frequency components is found based on prioritizing. Finally the steps of controlling a post-filtering of the transform in dependence of the finding, and inverse transforming the post-filtered transform in order to obtain a post-filtered decoded speech signal in the time domain are performed.
    • 用于在时域中输出解码语音信号的语音解码器的后处理方法基于具有非谐波和噪声缺陷的频谱提供高频分辨率。 这是通过使用高频分辨率变换(FFT)将解码的时域信号变换为频域信号而获得的。 然后在其频率区域(4kHz)中对频域信号的能量分布进行分析,以找到干扰频率分量,并对位于频谱的较高部分中的这种频率分量进行优先排序。 接下来,基于优先级来找到干扰频率分量的抑制度。 最后,执行根据发现来控制变换的后置滤波的步骤,以及逆时变逆变换以便在时域中获得后滤波的解码语音信号。
    • 4. 发明授权
    • Speech coding including soft adaptability feature
    • 语音编码包括软适应性特征
    • US06564183B1
    • 2003-05-13
    • US09469258
    • 1999-12-22
    • Roar HagenErik Ekudden
    • Roar HagenErik Ekudden
    • G10L1900
    • G10L19/12G10L19/002G10L19/18G10L2019/0008
    • A speech encoding/decoding apparatus. A speech encoding apparatus has a coding portion for receiving input information related to an uncoded signal representative of an original speech signal, the coding portion including a fixed coding portion for receiving the input information and producing a first coded signal estimate, and an adaptive coding portion for receiving the input information and producing a second coded signal estimate. A controller is connected to the fixed coding portion and the adaptive coding portion for receiving information indicative of speech characteristics of the uncoded signal and generates a control signal; and a code modifier receives the first coded signal estimate from the fixed coding portion and the control signal from the controller and produces a modified signal estimate.
    • 语音编码/解码装置。 语音编码装置具有编码部分,用于接收与代表原始语音信号的未编码信号有关的输入信息,编码部分包括用于接收输入信息并产生第一编码信号估计的固定编码部分,以及自适应编码部分 用于接收输入信息并产生第二编码信号估计。 控制器连接到固定编码部分和自适应编码部分,用于接收指示未编码信号的语音特性的信息,并产生控制信号; 代码修改器接收来自固定编码部分的第一编码信号估计和来自控制器的控制信号,并产生修改的信号估计。
    • 9. 发明授权
    • Distance and distortion estimation method and apparatus in channel optimized vector quantization
    • 通道优化矢量量化中的距离和失真估计方法及装置
    • US06321193B1
    • 2001-11-20
    • US09238386
    • 1999-01-27
    • Johan NyströmTomas SvenssonRoar HagenTor Björn Minde
    • Johan NyströmTomas SvenssonRoar HagenTor Björn Minde
    • G10L1100
    • H03M7/3082
    • A channel optimized vector quantization apparatus includes a device for weighting a sample vector x by a weighting matrix A and a device for weighting a set of code book vectors ĉr by a weighting matrix B. Device form a set of distance measures {dw(Ax,Bĉr)} representing the distance between the weighted sample vector Ax and each weighted code book vector Bĉr. Other device form a set of distortion measures {&agr;i(x)} by multiplying each distance measure by a channel transition probability Pr|i that an index r has been received at a decoder when an index i has been sent from an encoder and adding together these multiplied distance measures for each possible index r. Finally device determine an index imin corresponding to the smallest distortion measure &agr;i(x) and represents the sample vector by this index imin.
    • 信道优化矢量量化装置包括用于通过加权矩阵A对采样矢量x进行加权的装置,以及用于通过加权矩阵B对一组码本向量ĉr进行加权的装置。装置形成一组距离度量{dw(Ax, Bĉr)}表示加权采样矢量Ax与每个加权码本向量Bĉr之间的距离。 其他设备通过将每个距离测量乘以信道转换概率Pr | i来形成一组失真度量{alphai(x)},当索引i已经从编码器发送并且相加在一起时,在解码器处已经接收到索引r 对于每个可能的索引r,这些相乘的距离度量。 最后,设备确定对应于最小失真度量α(x)的索引值,并通过该索引imin表示样本向量。