公开(公告)号：US09135926B2

公开(公告)日：2015-09-15

申请号：US13613742

申请日：2012-09-13

Applicant: Mi-Suk Lee ,  Do-Young Kim ,  Byung-Sun Lee

Inventor： Mi-Suk Lee ,  Do-Young Kim ,  Byung-Sun Lee

IPC: G10L21/00 ,  G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G11B20/10 ,  H03G3/34 ,  G10L21/038

CPC classification number: G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G10L21/038 ,  G11B20/10527 ,  G11B2020/10555 ,  H03G3/341

Abstract: An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

公开(公告)号：US08751225B2

公开(公告)日：2014-06-10

申请号：US13106649

申请日：2011-05-12

Applicant: Mi-Suk Lee ,  Hong-Kook Kim ,  Young-Han Lee

Inventor： Mi-Suk Lee ,  Hong-Kook Kim ,  Young-Han Lee

IPC: G10L19/00

CPC classification number: G10L19/24 ,  G10L19/0208 ,  G10L19/0212 ,  G10L21/038

Abstract: Provided is an apparatus and method for encoding a voice and audio signal by expanding a modified discrete cosine transform (MDCT) based CODEC to a wideband and a super-wideband in a communication system. The apparatus for encoding a signal in a communication system, includes a converter configured to convert a time domain signal corresponding to a service to be provided to users to a frequency domain signal, a quantization and normalization unit configured to calculate and quantize gain of each subband in the converted frequency domain signal and normalize a frequency coefficient of the each subband, a search unit configured to search patch information of each subband in the converted frequency domain signal using the normalized frequency coefficient, and a packetizer configured to packetize the quantized gain and the searched patch information and encode gain information of each subband in the frequency domain signal.

Abstract translation: 提供了一种通过在通信系统中扩展基于修改的离散余弦变换（MDCT）的CODEC到宽带和超宽带来对语音和音频信号进行编码的装置和方法。 用于对通信系统中的信号进行编码的装置包括转换器，被配置为将与要提供给用户的服务相对应的时域信号转换为频域信号，量化和归一化单元，被配置为计算和量化每个子带的增益 在转换的频域信号中对归一化每个子带的频率系数，搜索单元，被配置为使用归一化的频率系数来搜索经转换的频域信号中的每个子带的贴片信息，以及分组器，被配置为将量化的增益和 搜索补丁信息，并对频域信号中的每个子带的增益信息进行编码。

公开(公告)号：US20110301961A1

公开(公告)日：2011-12-08

申请号：US13201517

申请日：2010-02-16

Applicant: Mi-Suk Lee ,  Hyun-Joo Bae ,  Byung-Sun Lee

Inventor： Mi-Suk Lee ,  Hyun-Joo Bae ,  Byung-Sun Lee

IPC: G10L21/00

CPC classification number: G10L19/008 ,  G10L19/0204 ,  G10L19/093 ,  G10L19/24

Abstract: A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently.

Abstract translation: 提供了一种使用自适应正弦编码对音频信号进行编码和解码的方法和装置。 音频信号编码方法包括以下步骤：将合成的音频信号分成多个子带，计算每个子带的能量，从子帧中选择具有相对大量能量的预定数量的子带， 并且对于所选择的子带执行正弦编码。 基于考虑合成信号的每个子带的能量量的正弦编码的应用更有效地提高了合成信号的质量。

公开(公告)号：US20110280337A1

公开(公告)日：2011-11-17

申请号：US13106649

申请日：2011-05-12

Applicant: Mi-Suk LEE ,  Hong-Kook KIM ,  Young-Han LEE

Inventor： Mi-Suk LEE ,  Hong-Kook KIM ,  Young-Han LEE

IPC: H04L27/00

CPC classification number: G10L19/24 ,  G10L19/0208 ,  G10L19/0212 ,  G10L21/038

Abstract: Provided is an apparatus and method for encoding a voice and audio signal by expanding a modified discrete cosine transform (MDCT) based CODEC to a wideband and a super-wideband in a communication system. The apparatus for encoding a signal in a communication system, includes a converter configured to convert a time domain signal corresponding to a service to be provided to users to a frequency domain signal, a quantization and normalization unit configured to calculate and quantize gain of each subband in the converted frequency domain signal and normalize a frequency coefficient of the each subband, a search unit configured to search patch information of each subband in the converted frequency domain signal using the normalized frequency coefficient, and a packetizer configured to packetize the quantized gain and the searched patch information and encode gain information of each subband in the frequency domain signal.

Abstract translation: 提供了一种通过在通信系统中扩展基于修改的离散余弦变换（MDCT）的CODEC到宽带和超宽带来对语音和音频信号进行编码的装置和方法。 用于对通信系统中的信号进行编码的装置包括转换器，被配置为将与要提供给用户的服务相对应的时域信号转换为频域信号，量化和归一化单元，被配置为计算和量化每个子带的增益 在转换的频域信号中对归一化每个子带的频率系数，搜索单元，被配置为使用归一化的频率系数来搜索经转换的频域信号中的每个子带的贴片信息，以及分组器，被配置为将量化的增益和 搜索补丁信息，并对频域信号中的每个子带的增益信息进行编码。

公开(公告)号：US07457744B2

公开(公告)日：2008-11-25

申请号：US10628058

申请日：2003-07-25

Applicant: Mi-suk Lee ,  Dae-hwan Hwang

Inventor： Mi-suk Lee ,  Dae-hwan Hwang

IPC: G10L11/04

CPC classification number: G10L25/90

Abstract: A device and a method for estimating an open-loop pitch in a general speech CODEC are disclosed. The open-loop pitch estimation device includes an autocorrelation function calculation unit which calculates a normalized autocorrelation function from a perceptual weighing filtered speech signal, a maximum autocorrelation function and lag estimation unit which estimates a maximum autocorrelation function and candidates for the maximum autocorrelation function, a pitch candidate decision unit which decides candidates for a pitch by using the ratio of the estimated maximum autocorrelation function to the candidates for the estimated maximum autocorrelation function, and lags of which values are smaller than a predetermined threshold value, and a pitch estimation unit which estimates a pitch between the candidates for a pitch and the lags corresponding to the estimated maximum autocorrelation function by using a pitch of a previous frame of the speech signal.

Abstract translation: 公开了一种用于估计通用语音编解码器中的开环音调的装置和方法。 开环音调估计装置包括自相关函数计算单元，该自相关函数计算单元从感知称重滤波语音信号，最大自相关函数和滞后估计单元计算归一化自相关函数，估计最大自相关函数和最大自相关函数候选， 音调候选决定单元，其通过使用估计的最大自相关函数与估计的最大自相关函数的候选的比率以及其值小于预定阈值的滞后来估计音调的候选，以及估计单元，其估计 通过使用语音信号的前一帧的音调，音调候选之间的间距和对应于估计的最大自相关函数的滞后。

公开(公告)号：US20060277040A1

公开(公告)日：2006-12-07

申请号：US11441955

申请日：2006-05-26

Applicant: Jong-Mo Sung ,  Hyun-Woo Kim ,  Mi-Suk Lee ,  Do-Young Kim

Inventor： Jong-Mo Sung ,  Hyun-Woo Kim ,  Mi-Suk Lee ,  Do-Young Kim

IPC: G10L19/00

CPC classification number: G10L21/038 ,  G10L19/0212 ,  G10L19/035 ,  G10L19/06 ,  G10L19/24

Abstract: Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, an LPC coefficient extractor, an LPC coefficient quantizer, an LP analysis filter, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The LPC coefficient extractor extracts LPC coefficients from the transform coefficients. The LPC coefficient quantizer quantizes the LPC coefficients to output quantized LPC coefficients and corresponding indices. The LP analysis filter performs an LP analysis on the transform coefficients to output LP residual transform coefficients. The band splitter splits the LP residual transform coefficients into bands to output the LP residual transform coefficients. The pulse searcher searches the LP residual transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses. The pulse quantizer quantizes the parameters of the optimal pulses.

Abstract translation: 提供了一种残留信号编码/解码装置和方法。 残余信号编码装置包括变压器，LPC系数提取器，LPC系数量化器，LP分析滤波器，带分离器，脉冲搜索器和脉冲量化器。 变压器将时域残差信号变换为频域，输出变换系数。 LPC系数提取器从变换系数中提取LPC系数。 LPC系数量化器量化LPC系数以输出量化的LPC系数和相应的索引。 LP分析滤波器对变换系数执行LP分析以输出LP残差变换系数。 频带分离器将LP残差变换系数分解成频带以输出LP残差变换系数。 脉冲搜索器搜索各个频带的LP残差变换系数，以选择最佳脉冲和最佳脉冲的输出参数。 脉冲量化器对最佳脉冲的参数进行量化。

公开(公告)号：US20050108009A1

公开(公告)日：2005-05-19

申请号：US10967045

申请日：2004-10-14

Applicant: Mi-Suk Lee ,  Do-Young Kim ,  Hong-Kook Kim ,  Seungho Choi

Inventor： Mi-Suk Lee ,  Do-Young Kim ,  Hong-Kook Kim ,  Seungho Choi

IPC: G10L19/00 ,  G10L19/14 ,  G10L19/02

CPC classification number: G10L19/24

Abstract: Disclosed is an apparatus for coding of variable bitrate wideband speech and audio, comprising: a) a speech and audio divider for dividing signals inputted to a CODEC into speech or audio signals; b) a narrowband coder for performing narrowband coding, in the case the divided input signals are speech signals; c) a bitrate modifier for modifying a bitrate for coding of a low frequency band and a bitrate for coding of a high frequency band, in the case the divided input signals are audio signals; and d) a wideband coder for performing coding by the modified bitrate in the bitrate modifier.

Abstract translation: 公开了一种用于编码可变比特率宽带语音和音频的装置，包括：a）语音和音频分频器，用于将输入到CODEC的信号分成语音或音频信号; b）用于执行窄带编码的窄带编码器，在分割的输入信号是语音信号的情况下; c）用于修改用于编码低频带的比特率和用于编码高频带的比特率的比特率修改器，在分割的输入信号是音频信号的情况下; 以及d）宽带编码器，用于通过比特率修改器中的修改的比特率执行编码。

公开(公告)号：US09142222B2

公开(公告)日：2015-09-22

申请号：US13613792

申请日：2012-09-13

Applicant: Mi-Suk Lee ,  Do-Young Kim ,  Byung-Sun Lee

Inventor： Mi-Suk Lee ,  Do-Young Kim ,  Byung-Sun Lee

IPC: G10L21/00 ,  G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G11B20/10 ,  H03G3/34 ,  G10L21/038

CPC classification number: G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G10L21/038 ,  G11B20/10527 ,  G11B2020/10555 ,  H03G3/341

Abstract: An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

Abstract translation: 提供了一种提高语音编解码器质量的装置和方法。 在该方法中，计算由低频带编解码器解码的信号的第一能量，并且计算通过低频带增强模式解码的信号的第二能量。 然后，当第一能量小于第一阈值或小于第二能量与第二阈值的乘积时，解码信号的大小被缩放。 因此，减少相对于静音段的量化误差的产生。

公开(公告)号：US20130246068A1

公开(公告)日：2013-09-19

申请号：US13876768

申请日：2011-09-28

Applicant: Mi-Suk Lee

Inventor： Mi-Suk Lee

IPC: G10L19/005

CPC classification number: G10L19/005 ,  G10L19/09

Abstract: Disclosed are a method and apparatus for decoding a an audiospeech signal using an adaptive codebook update. The method for decoding speechan audio signal includes: receiving an N+1-th normal frame data that is a normal frame transmitted after an N-th frame that is a loss frame data loss; determining whether an adaptive codebook of a final subframe of the N-th frame is updated or notby using the N-th frame and the N+1-th frame; updating the adaptive codebook of the final subframe of the N-th frame by using athe pitch index of the N+1-the frame; and synthesizing an audio a speech signal of by using the N+1-th frame.

Abstract translation: 公开了一种使用自适应码本更新来解码听力信号的方法和装置。 解码语音音频信号的方法包括：接收作为丢失帧数据丢失的第N帧之后发送的正常帧的N + 1个正常帧数据; 通过使用第N帧和第N + 1帧来确定第N帧的最后子帧的自适应码本是否被更新; 通过使用N + 1个帧的音调索引来更新第N帧的最后子帧的自适应码本; 以及通过使用第N + 1帧来合成音频语音信号。

公开(公告)号：US20130073282A1

公开(公告)日：2013-03-21

申请号：US13613742

申请日：2012-09-13

Applicant: Mi-Suk LEE ,  Do-Young KIM ,  Byung-Sun LEE

Inventor： Mi-Suk LEE ,  Do-Young KIM ,  Byung-Sun LEE

IPC: G10L21/00

CPC classification number: G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G10L21/038 ,  G11B20/10527 ,  G11B2020/10555 ,  H03G3/341

Abstract: An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

Abstract translation: 提供了一种提高语音编解码器质量的装置和方法。 在该方法中，计算由低频带编解码器解码的信号的第一能量，并且计算通过低频带增强模式解码的信号的第二能量。 然后，当第一能量小于第一阈值或小于第二能量与第二阈值的乘积时，解码信号的大小被缩放。 因此，减少相对于静音段的量化误差的产生。