公开(公告)号：US20090210219A1

公开(公告)日：2009-08-20

申请号：US12420215

申请日：2009-04-08

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: G10L21/00

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

Abstract: Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, 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 band splitter splits the transform coefficients into bands to output the transform coefficients. The pulse searcher searches the 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，分束器，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。 频带分离器将变换系数分成频带以输出变换系数。 脉冲搜索器搜索各个频带的变换系数，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：US20090150143A1

公开(公告)日：2009-06-11

申请号：US12155542

申请日：2008-06-05

Applicant: Hyun-woo Kim ,  Jong-mo Sung ,  Mi-suk Lee ,  Do-young Kim ,  Byung-sun Lee

Inventor： Hyun-woo Kim ,  Jong-mo Sung ,  Mi-suk Lee ,  Do-young Kim ,  Byung-sun Lee

IPC: G10L21/00 ,  G10L19/00

CPC classification number: G10L19/26 ,  G10L19/0212

Abstract: A post-filtering apparatus and method for speech enhancement in a modified discrete cosine transform (MDCT) domain are disclosed. In the apparatus and method, previous and current MDCT coefficients are used for obtaining a speech spectrum coefficient similar to a real speech spectrum, and a convex function is used for transforming the speech spectrum coefficient and obtaining a post-filter coefficient so that difference can increase in the case where the speech spectrum coefficient is small but decrease in the case where the coefficient is large. Then, the post-filter coefficient is applied to the MDCT coefficient. With this configuration, both the current and previous MDCT values are used, so that it is possible to obtain a spectrum coefficient similar to the real speech spectrum and to obtain a more accurate filter coefficient. Further, the coefficient is adaptively transformed through the convex function, thereby enhancing speech quality.

Abstract translation: 公开了一种用于修正的离散余弦变换（MDCT）域中的语音增强的后置滤波装置和方法。 在该装置和方法中，先前和当前的MDCT系数被用于获得类似于真实语音频谱的语音频谱系数，并且使用凸函数来变换语音频谱系数并获得后置滤波器系数，使得差值可以增加 在语音频谱系数小的情况下，在系数大的情况下减少。 然后，将后滤波器系数应用于MDCT系数。 利用该配置，使用当前和先前的MDCT值，使得可以获得与真实语音频谱相似的频谱系数并获得更准确的滤波器系数。 此外，系数通过凸函数自适应地变换，从而提高语音质量。

公开(公告)号：US20050049853A1

公开(公告)日：2005-03-03

申请号：US10932397

申请日：2004-09-01

Applicant: Mi-Suk Lee ,  Eung-Don Lee ,  Do-Young Kim ,  Hong-Kook Kim ,  Seung-Ho Choi

Inventor： Mi-Suk Lee ,  Eung-Don Lee ,  Do-Young Kim ,  Hong-Kook Kim ,  Seung-Ho Choi

IPC: G10L19/00

CPC classification number: G10L19/005

Abstract: Disclosed is a packet (frame) loss concealment method and device for a VoIP system, for reducing speech quality degradation caused by packet loss generated when transmitting speech data through a packet network. When a packet loss occur, the speech signal of the lost frame are reconstruct by combing the forward and backward prediction from the good frame before and after the lost frame Thus, speech quality of a speech coder can be improved without any extra delay in packet loss condition.

Abstract translation: 公开了一种用于VoIP系统的分组（帧）丢失隐藏方法和装置，用于在通过分组网络发送语音数据时减少由分组丢失引起的语音质量劣化。 当发生分组丢失时，通过从丢失帧之前和之后的良好帧组合前向和后向预测来重构丢失帧的语音信号。因此，可以改善语音编码器的语音质量，而不会在分组丢失中产生任何额外的延迟 条件。

公开(公告)号：US09135925B2

公开(公告)日：2015-09-15

申请号：US12529239

申请日：2008-11-28

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 core 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，计算由核心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：US08805680B2

公开(公告)日：2014-08-12

申请号：US13321297

申请日：2010-05-19

Applicant: Mi-Suk Lee ,  Heesik Yang ,  Hyun-Woo Kim ,  Jongmo Sung ,  Hyun-Joo Bae ,  Byung-Sun Lee

Inventor： Mi-Suk Lee ,  Heesik Yang ,  Hyun-Woo Kim ,  Jongmo Sung ,  Hyun-Joo Bae ,  Byung-Sun Lee

IPC: G10L19/02

CPC classification number: G10L19/02 ,  G10L19/0212 ,  G10L19/24

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract translation: 提供了一种用于对音频信号进行编码和解码的方法和装置。 一种对音频信号进行编码的方法包括接收经变换的音频信号，将变换的音频信号分成多个子带，对子带执行第一正弦脉冲编码操作，确定子带中的第二正弦脉冲编码操作的性能区域 基于第一正弦脉冲编码操作的编码信息，并对所确定的执行区域执行第二正弦脉冲编码操作，其中根据编码信息可变地执行第一正弦脉冲编码操作。 因此，通过考虑在通过分层正弦脉冲编码方案对上层中的音频信号进行编码或解码时下层的正弦脉冲编码，可以进一步提高合成信号的质量。

公开(公告)号：US08612239B2

公开(公告)日：2013-12-17

申请号：US12518263

申请日：2007-12-07

Applicant: Mi-Suk Lee ,  Do-Young Kim ,  Hae-Won Jung

Inventor： Mi-Suk Lee ,  Do-Young Kim ,  Hae-Won Jung

IPC: G10L21/00

CPC classification number: G10L19/008

Abstract: Provided is an audio coding apparatus and method that can selectively apply a operation mode of a coding module for stereo or multi-channel representation according to input signal characteristics of each channel, when voice or music signals are transmitted using an audio codec in portable terminals capable of stereo or multi-channel input and output. The audio coding apparatus includes a down-mixer for down-mixing multi-channel audio signals into mono signals; a coder for coding the mono signals; an input channel correlation analyzer for deciding whether to give them stereo effect based on their signal distribution characteristics, and outputting a control signal indicating whether to perform stereo representation process; and a stereo representation unit for performing stereo representation process onto the multi-channel audio signals when the control signal indicating to perform stereo representation process.

Abstract translation: 提供了一种音频编码装置和方法，当使用便携式终端的音频编解码器传输语音或音乐信号时，可以根据每个信道的输入信号特性选择性地应用用于立体声或多声道表示的编码模块的操作模式 的立体声或多声道输入和输出。 音频编码装置包括用于将多声道音频信号降混为单声道信号的下混频器; 用于对单声道信号进行编码的编码器; 输入信道相关分析器，用于基于它们的信号分布特性决定是否给出立体声效果，并输出指示是否执行立体声表示处理的控制信号; 以及立体声表示单元，用于当指示执行立体声表示处理的控制信号时，对多声道音频信号执行立体声表示处理。

公开(公告)号：US07773633B2

公开(公告)日：2010-08-10

申请号：US11635512

申请日：2006-12-08

Applicant: Mi Suk Lee ,  Dae Ho Kim ,  Do Young Kim

Inventor： Mi Suk Lee ,  Dae Ho Kim ,  Do Young Kim

IPC: H04J3/00

CPC classification number: H04L47/10 ,  H04L29/06027 ,  H04L47/2416 ,  H04L47/34 ,  H04L65/604 ,  H04L65/607 ,  H04L65/80

Abstract: Provided are a packet processing apparatus and method used when an output bitstream of an embedded codec is divided into a plurality of packets and transmitted accordingly, and more particularly, an efficient packet processing apparatus and method which can reduce deterioration of sound quality which may occur when a packet required for the reproduction of a voice signal is not received due to different arrival times of a plurality of packets at a receiving end if an output bitstream of an embedded codec is divided into the packets and transmitted accordingly through a path or a plurality of paths. In particular, an apparatus for processing packets of an embedded codec is provided. The apparatus includes a packet reception unit receiving packets, a layer information unit identifying layer information from received packets, a bitstream determination unit using the received packets, and a bitstream generation unit generating a new bitstream. The apparatus is used to provide a voice and multimedia service using an embedded codec in a packet network.

Abstract translation: 提供了当嵌入式编解码器的输出比特流被分成多个分组并相应地发送时使用的分组处理装置和方法，更具体地说，提供一种可以降低可能发生的声音质量恶化的有效分组处理装置和方法， 如果嵌入式编解码器的输出比特流被划分成分组并通过路径或多个分组相应地传送，则由于接收端的多个分组的到达时间不同，则不接收再现语音信号所需的分组 路径。 特别地，提供了一种用于处理嵌入式编解码器的分组的装置。 该装置包括接收分组的分组接收单元，从接收分组识别层信息的层信息单元，使用接收到的分组的比特流确定单元，以及生成新比特流的比特流生成单元。 该装置用于在分组网络中使用嵌入式编解码器来提供语音和多媒体服务。

公开(公告)号：US20100057449A1

公开(公告)日：2010-03-04

申请号：US12529239

申请日：2008-11-28

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

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

IPC: G10L19/00

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 core 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，计算由核心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：US07634402B2

公开(公告)日：2009-12-15

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: G10L19/02

CPC classification number: G10L19/24

Abstract: An apparatus for coding of variable bitrate wideband speech and audio is described. The apparatus utilizes: 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 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: 描述了用于编码可变比特率宽带语音和音频的装置。 该装置使用：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：US20060122828A1

公开(公告)日：2006-06-08

申请号：US11243955

申请日：2005-10-04

Applicant: Mi-Suk Lee ,  Jong-Mo Sung ,  Do-Young Kim ,  Hyun-Woo Kim ,  Kyung-Tae Kim ,  Hong-Goo Kang ,  Dae-Hee Youn ,  Sung-Kyo Jung

Inventor： Mi-Suk Lee ,  Jong-Mo Sung ,  Do-Young Kim ,  Hyun-Woo Kim ,  Kyung-Tae Kim ,  Hong-Goo Kang ,  Dae-Hee Youn ,  Sung-Kyo Jung

IPC: G10L19/10

CPC classification number: G10L19/04 ,  G10L19/0204 ,  G10L19/07 ,  G10L19/12

Abstract: Provided is a highband coding apparatus and method for a wideband coding system. The coding apparatus and method can reduce a pre-echo phenomenon by encoding the highband based on lowband encoding information and Temporal Noise Shaping technique. A highband encoding apparatus includes: a domain converter for converting the domain of an input highband signal into a frequency domain; a linear prediction order determiner for determining a linear prediction order based on the lowband encoding information; a linear prediction analyzer for analyzing a highband signal of the frequency domain based on the determined linear prediction order to thereby generate a linear prediction coefficient; a linear prediction coefficient quantizer for quantizing the linear prediction coefficient based on the lowband encoding information; and a residual signal quantizer for obtaining a residual signal by dequantizing the quantized linear prediction coefficient and quantizing the residual signal.

Abstract translation: 提供了一种用于宽带编码系统的高带编码装置和方法。 编码装置和方法可以通过基于低带编码信息和时间噪声成形技术对高频带进行编码来减少预回波现象。 高频编码装置包括：域转换器，用于将输入高频带信号的域转换成频域; 线性预测顺序确定器，用于基于所述低频带编码信息确定线性预测顺序; 线性预测分析器，用于基于所确定的线性预测顺序来分析频域的高频带信号，从而生成线性预测系数; 线性预测系数量化器，用于基于所述低频带编码信息量化所述线性预测系数; 以及残差信号量化器，用于通过对量化的线性预测系数进行去量化并量化残差信号来获得残留信号。