公开(公告)号：US20070276655A1

公开(公告)日：2007-11-29

申请号：US11709255

申请日：2007-02-22

申请人： Kangeun Lee ,  Eunmi Oh ,  Hosang Sung ,  Changyong Son ,  Kihyun Choo ,  Junghoe Kim

发明人： Kangeun Lee ,  Eunmi Oh ,  Hosang Sung ,  Changyong Son ,  Kihyun Choo ,  Junghoe Kim

IPC分类号： G06F15/00 ,  G06F17/28 ,  G11B11/00

CPC分类号： G10L19/12 ,  G10L19/24 ,  G10L2019/0013

摘要： A method and an apparatus to encode and decode a speech signal using a code excited linear prediction (CELP) algorithm. In order to reduce a bit rate without degrading performance in an enhancement layer based on CELP, each of a fixed codebook of a core layer and a fixed codebook of the enhancement layer is divided into a plurality of spaces. The spaces of the fixed codebook of the enhancement layer excludes a space corresponding to a least distorted space determined from among the spaces of the fixed codebook of the core layer are searched.

摘要翻译： 一种使用码激励线性预测（CELP）算法对语音信号进行编码和解码的方法和装置。 为了在基于CELP的增强层中降低比特率而不降低性能的情况下，核心层的固定码本和增强层的固定码本中的每一个被分成多个空格。 增强层的固定码本的空间不包括与核心层的固定码本的空格中确定的最小失真空间相对应的空间。

公开(公告)号：US20070016410A1

公开(公告)日：2007-01-18

申请号：US11457251

申请日：2006-07-13

申请人： Hosang Sung ,  Kungeun Lee ,  Sang-won Kang ,  Thomas Fischer ,  Ja-kyoung Jun

发明人： Hosang Sung ,  Kungeun Lee ,  Sang-won Kang ,  Thomas Fischer ,  Ja-kyoung Jun

IPC分类号： G10L19/12

CPC分类号： G10L19/0212 ,  G10L15/08 ,  G10L19/10 ,  G10L19/107 ,  G10L19/12 ,  G10L2019/0013

摘要： A method and apparatus to search a codebook including pulses that model a predetermined component of a speech signal. The method includes the operations of selecting a predetermined number of paths corresponding to a predetermined number of pulse locations that are most consistent with the predetermined component, from among paths corresponding to pulse locations of a predetermined pulse location set allocated to at least one branch that connects one state of a predetermined Trellis structure to another state, performing the path selecting operation on each of states other than the one state, and selecting a path corresponding to pulse locations that are most consistent with the predetermined component, from among paths including the selected paths. Accordingly, a number of calculations required during a codebook search is reduced.

摘要翻译： 一种搜索包括对语音信号的预定分量进行建模的脉冲的码本的方法和装置。 该方法包括从对应于分配给连接至少一个分支的预定脉冲位置集的脉冲位置的路径中选择对应于与预定分量最一致的预定数量的脉冲位置的预定数量的路径的操作 将预定网格结构的一个状态转换到另一状态，对除了一个状态以外的状态执行路径选择操作，并且从包括所选择的路径的路径中选择对应于与预定分量最一致的脉冲位置的路径 。 因此，减少了码本搜索期间所需的一些计算。

公开(公告)号：US20060217975A1

公开(公告)日：2006-09-28

申请号：US11337487

申请日：2006-01-24

申请人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

发明人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

IPC分类号： G10L21/02

CPC分类号： G10L19/0208 ,  G10L19/093

摘要： Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

摘要翻译： 在执行宽带音频编码和解码时使用高频带音频信号或宽带误差音频信号的谐波信息支持精细粒度可伸缩性（FGS）的音频编码和解码装置和方法，以及存储方法的记录介质。 音频编码方法包括检测输入音频信号的高频带音频信号或宽带误差音频信号的谐波; 确定检测到的谐波的顺序; 并根据确定的顺序对检测到的谐波进行编码。

公开(公告)号：US08473284B2

公开(公告)日：2013-06-25

申请号：US11097319

申请日：2005-04-04

申请人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

发明人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

IPC分类号： G10L19/12

CPC分类号： G10L19/07

摘要： A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.

摘要翻译： 语音编码/解码方法和装置。 语音编码器包括：量化选择单元，生成量化选择信号; 以及量化单元，从输入信号提取线性预测编码（LPC）系数，将所提取的LPC系数转换为线谱频率（LSF），基于所述线频谱频率（LSF）对第一LSF量化单元或第二LSF量化单元量化LSF 量化选择信号，并将量化的LSF转换成量化的LPC系数。 量化选择信号基于输入信号的先前帧中的合成语音信号的特性来选择第一LSF量化单元或第二LSF量化单元。

公开(公告)号：US08271267B2

公开(公告)日：2012-09-18

申请号：US11490139

申请日：2006-07-21

申请人： Hosang Sung ,  Sangwook Kim ,  Rakesh Taori ,  Kangeun Lee

发明人： Hosang Sung ,  Sangwook Kim ,  Rakesh Taori ,  Kangeun Lee

IPC分类号： G10L21/00

CPC分类号： G10L19/24 ,  G10L19/0212 ,  G10L25/18

摘要： Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.

摘要翻译： 提供了一种可扩展的宽带语音编码/解码装置，方法和媒体。 输入宽带语音输入信号首先被分成低频带信号和高频带信号。 然后使用码激励线性预测（CELP）方法对分频的低频带信号进行编码。 分频高频信号采用谐波法编码。 然后，使用修正的离散余弦变换（MDCT）方法对表示从低频带和高频带获得的合成信号之间的差异以及输入到低频带和高频带的信号进行编码的信号。 然后对编码信号进行多路复用。 然后输出复用的信号。 因此，可以实现对所有层的高质量语音。

公开(公告)号：US08015017B2

公开(公告)日：2011-09-06

申请号：US11337487

申请日：2006-01-24

申请人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

发明人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

IPC分类号： G10L19/00

CPC分类号： G10L19/0208 ,  G10L19/093

摘要： Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

摘要翻译： 在执行宽带音频编码和解码时使用高频带音频信号或宽带误差音频信号的谐波信息支持精细粒度可伸缩性（FGS）的音频编码和解码装置和方法，以及存储方法的记录介质。 音频编码方法包括检测输入音频信号的高频带音频信号或宽带误差音频信号的谐波; 确定检测到的谐波的顺序; 并根据确定的顺序对检测到的谐波进行编码。

公开(公告)号：US07765100B2

公开(公告)日：2010-07-27

申请号：US11347429

申请日：2006-02-06

申请人： Hosang Sung ,  Seungho Choi ,  Kihyun Choo

发明人： Hosang Sung ,  Seungho Choi ,  Kihyun Choo

IPC分类号： G10L11/00 ,  G10L19/14 ,  G10L13/00 ,  G10L13/06 ,  G10L19/00

CPC分类号： G10L19/005 ,  G10L19/07

摘要： A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame. The method and apparatus can improve the quality of a recovered speech signal, be applied to a variety of technologies, and provide a method of recovering an LSP parameter for development of an algorithm for speech decoding.

公开(公告)号：US20070040709A1

公开(公告)日：2007-02-22

申请号：US11485468

申请日：2006-07-13

申请人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee ,  Shihwa Lee ,  Sangwook Kim ,  Miyoung Kim ,  Dohyung Kim

发明人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee ,  Shihwa Lee ,  Sangwook Kim ,  Miyoung Kim ,  Dohyung Kim

IPC分类号： H03M7/00

CPC分类号： G10L19/0208

摘要： A method and apparatus to scalably encode and/or decode an audio signal includes encoding a specific band signal included in an input signal, encoding a frequency envelope of an excited signal in which the encoded specific band signal is removed from the input signal, encoding a residual signal in which the encoded frequency envelope is removed from the excited signal, and forming a bit-stream by scalably packing the encoded specific band signal, frequency envelop, and residual signal.

摘要翻译： 一种用于对音频信号进行可缩放编码和/或解码的方法和装置包括编码包括在输入信号中的特定频带信号，对其中去除编码的特定频带信号的激励信号的频率包络进行编码， 残留信号，其中编码的频率包络从激励信号中去除，以及通过可扩展地打包编码的特定频带信号，频率包络和残余信号来形成比特流。

公开(公告)号：US20060074643A1

公开(公告)日：2006-04-06

申请号：US11097319

申请日：2005-04-04

申请人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

发明人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

IPC分类号： G10L19/12

CPC分类号： G10L19/07

摘要： A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The the quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.

摘要翻译： 语音编码/解码方法和装置。 语音编码器包括：量化选择单元，生成量化选择信号; 以及量化单元，从输入信号提取线性预测编码（LPC）系数，将所提取的LPC系数转换为线谱频率（LSF），基于所述线频谱频率（LSF）对第一LSF量化单元或第二LSF量化单元量化LSF 量化选择信号，并将量化的LSF转换成量化的LPC系数。 量化选择信号基于输入信号的先前帧中的合成语音信号的特性来选择第一LSF量化单元或第二LSF量化单元。

公开(公告)号：US20060020453A1

公开(公告)日：2006-01-26

申请号：US11128432

申请日：2005-05-13

申请人： Changyong Son ,  Hosang Sung ,  Hochong Park ,  Byounghak Jeong ,  Youngyo Kim

发明人： Changyong Son ,  Hosang Sung ,  Hochong Park ,  Byounghak Jeong ,  Youngyo Kim

IPC分类号： G10L19/00

CPC分类号： G10L19/025

摘要： A speech signal compression and/or decompression method, medium, and apparatus in which the speech signal is transformed into the frequency domain for quantizing and dequantizing information of frequency coefficients. The speech signal compression apparatus includes a transform unit to transform a speech signal into the frequency domain and obtain frequency coefficients, a magnitude quantization unit to transform magnitudes of the frequency coefficients, quantize the transformed magnitudes and obtain magnitude quantization indices, a sign quantization unit to quantize signs of the frequency coefficients and obtain sign quantization indices, and a packetizing unit to generate the magnitude and sign quantization indices as a speech packet.

摘要翻译： 一种语音信号压缩和/或解压缩方法，介质和装置，其中语音信号被变换成频域以量化和去量化频率系数的信息。 语音信号压缩装置包括将语音信号变换为频域并获得频率系数的变换单元，变换频率系数的幅度量化单位，量化变换幅度并获得幅度量化索引，符号量化单元， 量化频率系数的符号并获得符号量化索引，以及分组单元，用于生成幅度和符号量化索引作为语音分组。