公开(公告)号：US20080312917A1

公开(公告)日：2008-12-18

申请号：US12190524

申请日：2008-08-12

申请人： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. Dejaco

发明人： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. Dejaco

IPC分类号： G10L19/00

CPC分类号： G10L19/04 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/08 ,  G10L19/097 ,  G10L19/26 ,  G10L25/12

摘要： A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.

摘要翻译： 用于预测量化浊音的方法和装置包括参数发生器和量化器。 参数发生器被配置为从诸如有声语音的预测语音的帧中提取参数，并且将提取的信息变换为频域表示。 量化器被配置为从当前帧的参数中减去先前帧的参数的加权和。 量化器被配置为量化差值。 可以添加原型提取器以首先提取要由参数发生器处理的音调周期原型。

公开(公告)号：US07426466B2

公开(公告)日：2008-09-16

申请号：US10897746

申请日：2004-07-22

申请人： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. DeJaco

发明人： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. DeJaco

IPC分类号： G10L19/14

CPC分类号： G10L19/04 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/08 ,  G10L19/097 ,  G10L19/26 ,  G10L25/12

摘要： A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.

摘要翻译： 用于预测量化浊音的方法和装置包括参数发生器和量化器。 参数发生器被配置为从诸如有声语音的预测语音的帧中提取参数，并且将提取的信息变换为频域表示。 量化器被配置为从当前帧的参数中减去先前帧的参数的加权和。 量化器被配置为量化差值。 可以添加原型提取器以首先提取要由参数发生器处理的音调周期原型。

公开(公告)号：US06434519B1

公开(公告)日：2002-08-13

申请号：US09356861

申请日：1999-07-19

申请人： Sharath Manjunath ,  Andrew P. Dejaco ,  Arasanipalai K. Ananthapadmanabhan ,  Pengjun Huang ,  Eddie Lun Tik Choy

发明人： Sharath Manjunath ,  Andrew P. Dejaco ,  Arasanipalai K. Ananthapadmanabhan ,  Pengjun Huang ,  Eddie Lun Tik Choy

IPC分类号： G10L1914

CPC分类号： G10L19/0208 ,  G10L19/10

摘要： A method and apparatus for identifying frequency bands to compute linear phase shifts between frame prototypes in a speech coder includes partitioning the frequency spectrum of a prototype of a frame by dividing the frequency spectrum into segments, assigning one or more bands to each segment, and establishing, for each segment, a set of bandwidths for the bands. The bandwidths may be fixed and uniformly distributed in any given segment. The bandwidths may be fixed and non-uniformly distributed in any segment. The bandwidths may be variable and non-uniformly distributed in any given segment.

摘要翻译： 用于识别用于计算语音编码器中的帧原型之间的线性相移的频带的方法和装置包括：通过将频谱划分成段，将一个或多个频带分配给每个分段来建立帧的原型的频谱，并建立 ，对于每个段，一组带宽的带宽。 带宽可以是固定的，并且均匀分布在任何给定的段中。 带宽可以是固定的，并且不均匀地分布在任何段中。 带宽可以是可变的，并且不均匀地分布在任何给定的段中。

公开(公告)号：US06438518B1

公开(公告)日：2002-08-20

申请号：US09429754

申请日：1999-10-28

申请人： Sharath Manjunath ,  Andrew P. Dejaco ,  Arasanipalai K. Ananthapadmanabhan ,  Eddie Lun Tik Choy

发明人： Sharath Manjunath ,  Andrew P. Dejaco ,  Arasanipalai K. Ananthapadmanabhan ,  Eddie Lun Tik Choy

IPC分类号： G10L1904

CPC分类号： G10L19/18 ,  G10L19/02

摘要： A method and apparatus for using coding scheme selection patterns in a predictive speech coder to reduce sensitivity to frame error conditions includes a speech coder configured to select from among various predictive coding modes. After a predefined number of speech frames have been predictively coded, the speech coder codes one frame with a nonpredictive coding mode or a mildly predictive coding mode. The predefined number of frames can be determined in advance from the subjective standpoint of a listener. The predefined number of frames may be varied periodically. An average coding bit rate may be maintained for the speech coder by ensuring that an average coding bit rate is maintained for each successive pattern, or group, of predictively coded speech frames including at least one nonpredictively coded or mildly predictively coded speech frame.

摘要翻译： 一种用于在预测语音编码器中使用编码方案选择模式以降低对帧错误状况的灵敏度的方法和装置包括配置成从各种预测编码模式中进行选择的语音编码器。 在预定数量的语音帧已被预测编码之后，语音编码器以非预测编码模式或轻度预测编码模式对一帧进行编码。 可以从收听者的主观角度预先确定预定数量的帧。 预定数量的帧可以周期性地改变。 可以通过确保针对包括至少一个非预测编码或温和预测编码的语音帧的预测编码语音帧的每个连续模式或组维持平均编码比特率来维持语音编码器的平均编码比特率。

公开(公告)号：US06584438B1

公开(公告)日：2003-06-24

申请号：US09557283

申请日：2000-04-24

申请人： Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy

发明人： Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy

IPC分类号： G10L1300

CPC分类号： G10L21/02 ,  G10L19/005 ,  G10L19/097

摘要： A frame erasure compensation method in a variable-rate speech coder includes quantizing, with a first encoder, a pitch lag value for a current frame and a first delta pitch lag value equal to the difference between the pitch lag value for the current frame and the pitch lag value for the previous frame. A second, predictive encoder quantizes only a second delta pitch lag value for the previous frame (equal to the difference between the pitch lag value for the previous frame and the pitch lag value for the frame prior to that frame). If the frame prior to the previous frame is processed as a frame erasure, the pitch lag value for the previous frame is obtained by subtracting the first delta pitch lag value from the pitch lag value for the current frame. The pitch lag value for the erasure frame is then obtained by subtracting the second delta pitch lag value from the pitch lag value for the previous frame. Additionally, a waveform interpolation method may be used to smooth discontinuities caused by changes in the coder pitch memory.

摘要翻译： 可变速率语音编码器中的帧擦除补偿方法包括：利用第一编码器量化当前帧的音调滞后值，以及等于当前帧的音调滞后值与第 前一帧的音调滞后值。 第二预测编码器仅量化前一帧的第二增量音调滞后值（等于先前帧的音调滞后值与该帧之前的帧的音调滞后值之间的差）。 如果先前帧之前的帧被作为帧擦除处理，则通过从当前帧的音调滞后值中减去第一增量音调滞后值来获得先前帧的音调滞后值。 然后通过从前一帧的音调滞后值减去第二增量音调滞后值来获得擦除帧的音调滞后值。 此外，可以使用波形插值方法来平滑由编码器音调存储器的变化引起的不连续性。

公开(公告)号：US06324505B1

公开(公告)日：2001-11-27

申请号：US09356756

申请日：1999-07-19

申请人： Eddie Lun Tik Choy ,  Sharath Manjunath

发明人： Eddie Lun Tik Choy ,  Sharath Manjunath

IPC分类号： G10L2102

CPC分类号： G10L19/0204 ,  G10L25/18

摘要： An amplitude quantization scheme for low-bit-rate speech coders includes the first step of extracting a vector of spectral information from a frame. The energy of the vector is normalized to generate gain factors. The gain factors are differentially vector quantized. The normalized gain factors are non-uniformly downsampled to generate a fixed-dimension vector with elements associated with a set of non-uniform frequency bands. The fixed-dimension vector is split into two or more sub-vectors. The sub-vectors are differentially quantized, to best advantage with a harmonic cloning process.

摘要翻译： 用于低比特率语音编码器的幅度量化方案包括从帧提取频谱信息的向量的第一步骤。 向量的能量被归一化以产生增益因子。 增益因子是差分矢量量化的。 归一化的增益因子被非均匀地下采样以产生具有与一组非均匀频带相关联的元素的固定维度向量。 固定维度向量被分成两个或多个子向量。 子矢量被差分量化，以利用谐波克隆过程的最佳优势。

公开(公告)号：US06393394B1

公开(公告)日：2002-05-21

申请号：US09356755

申请日：1999-07-19

申请人： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath

发明人： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath

IPC分类号： G10L2100

CPC分类号： G10L19/07 ,  G10L25/12 ,  G10L2019/0005

摘要： A method and apparatus for interleaving line spectral information quantization methods in a speech coder includes quantizing line spectral information with two vector quantization techniques, the first technique being a non-moving-average prediction-based technique, and the second technique being a moving-average prediction-based technique. A line spectral information vector is vector quantized with the first technique. Equivalent moving average codevectors for the first technique are computed. A memory of a moving average codebook of codevectors is updated with the equivalent moving average codevectors for a predefined number of frames that were previously processed by the speech coder. A target quantization vector for the second technique is calculated based on the updated moving average codebook memory. The target quantization vector is vector quantized with the second technique to generate a quantized target codevector. The memory of the moving average codebook is updated with the quantized target codevector. Quantized line spectral information vectors are derived from the quantized target codevector.

摘要翻译： 用于在语音编码器中交织线谱信息量化方法的方法和装置包括使用两个矢量量化技术量化线谱信息，第一技术是基于非移动平均预测的技术，第二技术是移动平均 基于预测的技术。 线谱信息矢量用第一技术进行矢量量化。 计算第一种技术的等效移动平均码矢量。 代码矢量的移动平均码本的存储器用先前由语音编码器处理的预定数量的帧的等效移动平均码向量更新。 基于更新的移动平均码本存储器计算第二技术的目标量化矢量。 目标量化矢量用第二技术进行矢量量化，以产生量化的目标码矢量。 用量化的目标码矢量来更新移动平均码本的存储器。 量化的线谱信息矢量从量化的目标码矢量导出。

公开(公告)号：US06330532B1

公开(公告)日：2001-12-11

申请号：US09356493

申请日：1999-07-19

申请人： Sharath Manjunath ,  Andrew P. Dejaco

发明人： Sharath Manjunath ,  Andrew P. Dejaco

IPC分类号： G10L2104

CPC分类号： G10L19/002 ,  G10L19/18

摘要： A method and apparatus for maintaining a target bit rate in a speech coder includes a speech coder for encoding a frame at a preselected encoding rate, computing a running average bit rate for a predefined number of encoded frames, subtracting the running average bit rate from a predefined target average bit rate, and dividing the difference by the preselected encoding rate. If the quotient value is negative, a predefined number of possible occurrence counts of speech coder performance threshold values that are less than a current performance threshold value is accumulated, the accumulated number being greater than the absolute value of the quotient. The product of a decrement-per-occurrence-count-value and the predefined number of occurrence counts is subtracted from the current performance threshold value to obtain a new performance threshold value. If the quotient value is positive, a predefined number of possible occurrence counts of speech coder performance threshold values that are greater than the current performance threshold value is accumulated, the accumulated number being greater than the quotient. The product of an increment-per-occurrence-count-value and the predefined number of occurrence counts is added to the current performance threshold value to obtain a new performance.

摘要翻译： 用于在语音编码器中维持目标比特率的方法和装置包括语音编码器，用于以预先选择的编码速率对帧进行编码，计算预定数量编码帧的运行平均比特率，从 预定义的目标平均比特率，并且将差除以预选的编码率。 如果商值为负，则累积小于当前性能阈值的语音编码器性能阈值的预定数量的可能发生计数，累积数大于商的绝对值。 从当前性能阈值中减去每次出现计数值递减和预定发生次数的乘积，以获得新的性能阈值。 如果商值为正，则累积大于当前性能阈值的语音编码器性能阈值的预定数量的可能发生计数，累积数大于商。 将每个出现次数增量值和预定发生次数的乘积加到当前性能阈值以获得新的性能。

公开(公告)号：US06766289B2

公开(公告)日：2004-07-20

申请号：US09874657

申请日：2001-06-04

申请人： Ananthapadmanabhan Kandhadai ,  Andrew P. DeJaco ,  Sharath Manjunath

发明人： Ananthapadmanabhan Kandhadai ,  Andrew P. DeJaco ,  Sharath Manjunath

IPC分类号： G10L1910

CPC分类号： G10L19/10 ,  G10L2019/0013

摘要： Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. In encoding schemes that use forward and backward pitch enhancement, storage and processor load is reduced by approximating a two-dimensional autocorrelation matrix with a one-dimensional autocorrelation vector. The approximation is possible when a cross-correlation element is configured to determine the autocorrelation matrix of an impulse response and a pulse energy determination element is configured to determine the energy of a pulse code vector that incorporates secondary pulse positions.

摘要翻译： 本文给出了从码本快速选择最佳激励波形的方法和装置。 在使用前向和后向间距增强的编码方案中，通过用一维自相关向量逼近二维自相关矩阵来减少存储和处理器负载。 当互相关元件被配置为确定脉冲响应的自相关矩阵并且脉冲能量确定元件被配置为确定包含次级脉冲位置的脉冲码矢量的能量时，近似是可能的。

公开(公告)号：US06789059B2

公开(公告)日：2004-09-07

申请号：US09876352

申请日：2001-06-06

申请人： Ananthapadmanabhan Kandhadai ,  Andrew P. DeJaco ,  Sharath Manjunath

发明人： Ananthapadmanabhan Kandhadai ,  Andrew P. DeJaco ,  Sharath Manjunath

IPC分类号： G10L1910

CPC分类号： G10L19/10 ,  G10L2019/0013

摘要： Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. To reduce the number of computations required to choose the optimal codebook vector, a subset of codevectors are selected based upon optimal pulse locations, wherein the subset of codevectors form a subcodebook. Rather than searching the entire codebook, only the entries of the subcodebook are searched.

摘要翻译： 本文给出了从码本快速选择最佳激励波形的方法和装置。 为了减少选择最佳码本向量所需的计算次数，基于最佳脉冲位置选择码矢量的子集，其中码矢量子集形成子码本。 而不是搜索整个码本，只搜索子码本的条目。