公开(公告)号：US20080147414A1

公开(公告)日：2008-06-19

申请号：US11939074

申请日：2007-11-13

Applicant: Chang-yong Son ,  Eun-mi Oh ,  Ki-hyun Choo ,  Jung-hoe Kim ,  Ho-sang Sung ,  Kang-eun Lee

Inventor： Chang-yong Son ,  Eun-mi Oh ,  Ki-hyun Choo ,  Jung-hoe Kim ,  Ho-sang Sung ,  Kang-eun Lee

IPC: G10L19/06

CPC classification number: G10L19/20 ,  G10L25/78

Abstract: A method and apparatus to determine an encoding mode of an audio signal, and a method and apparatus to encode an audio signal according to the encoding mode. In the encoding mode determination method, a mode determination threshold for the current frame that is subject to encoding mode determination is adaptively adjusted according to a long-term feature of the audio signal for a frame (the current frame) that is subject to encoding mode determination, thereby improving the hit rate of encoding mode determination and signal classification, suppressing frequent oscillation of an encoding mode in frame units, improving noise tolerance, and improving smoothness of a reconstructed audio signal.

Abstract translation: 一种确定音频信号的编码模式的方法和装置，以及根据编码模式对音频信号进行编码的方法和装置。 在编码模式确定方法中，根据用于编码模式的帧（当前帧）的音频信号的长期特征，自适应地调整用于进行编码模式确定的当前帧的模式确定阈值 确定，从而提高编码模式确定和信号分类的命中率，以帧为单位抑制编码模式的频繁振荡，提高噪声容限并提高重构音频信号的平滑度。

公开(公告)号：US07389227B2

公开(公告)日：2008-06-17

申请号：US09749782

申请日：2000-12-28

Applicant: Sang Won Kang ,  Chang Yong Son ,  Won Il Lee ,  Yoo Na Sung ,  Min Kyu Shim ,  Seong Hoon Hong

Inventor： Sang Won Kang ,  Chang Yong Son ,  Won Il Lee ,  Yoo Na Sung ,  Min Kyu Shim ,  Seong Hoon Hong

IPC: G10L19/12

CPC classification number: G10L19/07 ,  G10L2019/0005

Abstract: A high-speed search method in a speech encoder using an order character of LSP (Line Spectrum Pair) parameters in an LSP parameter quantizer using SVQ (Split Vector Quantization) used in a low-speed transmission speech encoder, includes the steps of rearranging a codebook according to an element value of a reference row for determining a range of code vectors to be searched; and determining a search range by using an order character between a given target vector and an arranged code vector to obtain an optimal code vector. The method gives effects of reducing computational complexity required to search the codebook without signal distortion in quantizing the LSP parameters of the speech encoder using SVQ, and reducing computational complexity without loss of tone quality in G.729 fixed codebook search by performing candidate selection and search on the basis of the correlation value size of the pulse position index.

Abstract translation: 在使用在低速传输语音编码器中使用的SVQ（分割矢量量化）的LSP参数量化器中使用LSP（线谱对）参数的顺序字符的语音编码器中的高速搜索方法包括以下步骤： 码本，根据用于确定要搜索的码矢量的范围的参考行的元素值; 以及通过使用给定目标向量和排列的代码向量之间的顺序字符来确定搜索范围，以获得最佳代码向量。 该方法提供了在使用SVQ量化语音编码器的LSP参数的同时降低搜索码本所需的计算复杂度，并且通过执行候选者选择和搜索来降低计算复杂度而不损失G.729固定码本搜索中的音调质量的效果 基于脉冲位置指数的相关值大小。

公开(公告)号：US20080140428A1

公开(公告)日：2008-06-12

申请号：US11949925

申请日：2007-12-04

Applicant: Ki-hyun Choo ,  Eun-mi Oh ,  Chang-yong Son ,  Jung-hoe Kim

Inventor： Ki-hyun Choo ,  Eun-mi Oh ,  Chang-yong Son ,  Jung-hoe Kim

IPC: G10L19/00 ,  G10L19/02

CPC classification number: G10L19/022

Abstract: An apparatus to process an audio signal includes an encoder to determine a variable window size of sub bands of a frame of an audio signal, to transform the sub bands according to the variable window size from a first domain to a second domain, to quantize the transformed sub bands, and to multiplex the quantized sub bands and information on the variable window size corresponding the respective sub bands.

Abstract translation: 一种用于处理音频信号的装置包括：编码器，用于确定音频信号的帧的子频带的可变窗口大小，以便根据可变窗口尺寸从第一域到第二域变换子带，以量化 并且对量化的子带和关于对应于各个子带的可变窗口大小的信息进行多路复用。

公开(公告)号：US20070282599A1

公开(公告)日：2007-12-06

申请号：US11757528

申请日：2007-06-04

Applicant: Ki-hyun CHOO ,  Jung-hoe Kim ,  Eun-ml Oh ,  Mino Lei ,  Chang-yong Son

Inventor： Ki-hyun CHOO ,  Jung-hoe Kim ,  Eun-ml Oh ,  Mino Lei ,  Chang-yong Son

IPC: G10L19/14

CPC classification number: G10L19/0208 ,  G10L21/038

Abstract: A method and apparatus to perform bandwidth extension encoding and decoding encodes and/or decodes a high frequency signal using an excitation signal for a low frequency signal encoded in a time domain or a frequency domain or using an excitation spectrum for the low frequency signal. Accordingly, although an audio signal is encoded or decoded using a small number of bits, the quality of sound corresponding to a signal in a high frequency band does not degrade. Therefore, a coding efficiency of the audio signal can be maximized.

Abstract translation: 执行带宽扩展编码和解码的方法和装置使用用于在时域或频域中编码的低频信号的激励信号或者使用低频信号的激励频谱对高频信号进行编码和/或解码。 因此，尽管使用少量的比特对音频信号进行编码或解码，但是与高频带中的信号相对应的声音质量不会降低。 因此，可以使音频信号的编码效率最大化。

公开(公告)号：US07080007B2

公开(公告)日：2006-07-18

申请号：US10253418

申请日：2002-09-25

Applicant: Chang-yong Son ,  Vladimir Shin ,  Sang-ryong Kim

Inventor： Chang-yong Son ,  Vladimir Shin ,  Sang-ryong Kim

IPC: G10L11/02

CPC classification number: G10L25/78 ,  G10L21/02

Abstract: An apparatus and a method for computing a Speech Absence Probability (SAP), and an apparatus and a method for removing noise by using the SAP computing device and method are provided. The provided SAP computing device for computing the SAP indicating probability that speech is absent in a mth frame, from a first through Ncth posteriori (Nc means the total number of channels) Signal to Noise Ratios (SNR) calculated with regard to the mth frame of a speech signal and a first through Ncth predicted SNRs predicted with regard to the mth frame, includes: a first through Ncth likelihood ratio generators for generating a first through Ncth likelihood ratios from the first through Ncth posterior SNRs and the first through Ncth predicted SNRs, and outputting them; a first multiplying unit for multiplying the first through Ncth likelihood ratios by a predetermined a priori probability, and outputting the multiplication results; an adding unit for adding each of the multiplication results received from the first multiplying unit to a predetermined value, and outputting the added results; a second multiplying unit for multiplying the added results received from the adding unit and outputting the multiplication result; and a inverse number calculator for calculating inverse number of the multiplication result received from the second multiplying unit and outputting the calculated inverse number as the SAP. Therefore, since the accuracy of the calculated SAP is high, noise can be efficiently removed from the speech signal that may have noise and an enhanced speech signal with an enhanced quality can be provided.

Abstract translation: 提供了一种用于计算语音缺席概率（SAP）的装置和方法，以及通过使用SAP计算装置和方法来消除噪声的装置和方法。 提供的用于计算SAP的SAP计算设备，指示在第帧中从第一至Nc第（S）个后验（Nc表示总信道数）的语音不存在语音的概率， 相对于语音信号的第m帧和/或第一至第N个预测SNR计算的信噪比（SNR）与第m个/ 包括：第一至第N个第一个似然比发生器，用于产生从第一到第N个第个似然比， / SUP>后验SNR和第一至第N个第个预测SNR，并输出它们; 第一乘法单元，用于将第一至第N个第个似然比乘以预定的先验概率，并输出乘法结果; 加法单元，用于将从第一乘法单元接收的乘法结果中的每一个相加到预定值，并输出相加结果; 第二乘法单元，用于将从加法单元接收的相加结果相乘并输出相乘结果; 以及逆数计算器，用于计算从第二乘法单元接收的乘法结果的倒数，并输出计算出的倒数作为SAP。 因此，由于计算出的SAP的精度高，所以可以从可能具有噪声的语音信号中有效地去除噪声，并且可以提供具有提高质量的增强语音信号。

公开(公告)号：US20050010404A1

公开(公告)日：2005-01-13

申请号：US10886662

申请日：2004-07-09

Applicant: Chang-yong Son ,  Kang-eun Lee ,  Sang-won Kang ,  Sang-hyun Chi

Inventor： Chang-yong Son ,  Kang-eun Lee ,  Sang-won Kang ,  Sang-hyun Chi

IPC: G10L19/12 ,  G10L19/14 ,  H03M7/30 ,  G10L19/04

CPC classification number: G10L19/24 ,  G10L19/12

Abstract: A coding apparatus including a base layer, a speech quality enhancement layer, and a multiplexer. The base layer filters an input speech signal using linear prediction coding and generates an excitation signal corresponding to the filtered speech signal through a fixed codebook search and an adaptive codebook search. The speech quality enhancement layer searches a fixed codebook using parameters obtained through the fixed codebook search in the base layer, or searches the fixed codebook using a target signal, which is obtained by removing a contribution of a fixed codebook of the base layer and a signal which is obtained by synthesizing and filtering a previous fixed codebook of the speech quality enhancement layer from a target signal for the fixed codebook search of the base layer. The multiplexer multiplexes signals generated by the base layer and the at least one speech quality enhancement layer.

Abstract translation: 一种包括基本层，语音质量增强层和多路复用器的编码装置。 基层使用线性预测编码对输入语音信号进行滤波，并通过固定码本搜索和自适应码本搜索产生与经滤波的语音信号相对应的激励信号。 语音质量增强层使用在基层中通过固定码本搜索获得的参数来搜索固定码本，或者使用目标信号搜索固定码本，该目标信号是通过去除基本层的固定码本的贡献和信号 通过从基本层的固定码本搜索的目标信号合成和过滤语音质量增强层的先前固定码本而获得。 复用器复用由基本层和至少一个语音质量增强层产生的信号。

公开(公告)号：US20050004794A1

公开(公告)日：2005-01-06

申请号：US10882339

申请日：2004-07-02

Applicant: Chang-yong Son ,  Ho-chong Park ,  Yong-beom Lee ,  Woo-suk Lee

Inventor： Chang-yong Son ,  Ho-chong Park ,  Yong-beom Lee ,  Woo-suk Lee

IPC: G10L19/00 ,  G10L19/14 ,  H03M7/30 ,  G10L19/04

CPC classification number: G10L19/24

Abstract: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract translation: 一种语音压缩装置，包括：将宽带语音信号变换为窄带低频语音信号的第一频带变换单元; 窄带语音压缩器压缩窄带低频语音信号并输出​​压缩结果作为低频语音分组; 解压缩单元解压缩低频带语音分组并获得解压缩的宽带低频语音信号; 误差检测单元，检测对应于宽带语音信号和解压缩宽带低频带语音信号之间的差的误差信号; 以及高频语音压缩单元，压缩宽带语音信号的误差信号和高频带语音信号，并将压缩结果输出为高频带语音分组。

公开(公告)号：US06622120B1

公开(公告)日：2003-09-16

申请号：US09498998

申请日：2000-02-04

Applicant: Byung Sik Yoon ,  Sang Won Kang ,  Chang Yong Son ,  Hyoung Jung Kim ,  Jung Chul Lee

Inventor： Byung Sik Yoon ,  Sang Won Kang ,  Chang Yong Son ,  Hyoung Jung Kim ,  Jung Chul Lee

IPC: G10L1912

CPC classification number: G10L19/06

Abstract: A fast search method for LSP (Linear Spectrum Pair) quantization is provided. The fast search method in accordance with an embodiment of the present invention includes the following steps. A first step is obtaining a target vector and a code vector. The target vector and the code vector are converted for ordering property. A second step is generating a code book having the ordering property for sub-matrices by utilizing the target vector and the code vector. A third step is selecting a particular line for determining a search scope in the code books and sorting the code book in descending order with respect to component values of the particular line. A fourth step is determining the search scope by utilizing the ordering property of the target vector and the sorted code vectors. The fifth step is obtaining an error standard by utilizing the target vector and the code vector, and obtaining an optimal code vector by utilizing the error standard within the determined search scope.

Abstract translation: 提供了一种用于LSP（线性频谱对）量化的快速搜索方法。 根据本发明的实施例的快速搜索方法包括以下步骤。 第一步是获得目标矢量和码矢量。 目标向量和代码矢量被转换为排序属性。 第二步是利用目标矢量和码矢量生成具有子矩阵的排序特性的码本。 第三步是选择用于确定代码簿中的搜索范围的特定行，并且相对于特定行的分量值按降序排序代码本。 第四步是通过利用目标矢量的排序属性和分类的代码矢量来确定搜索范围。 第五步是通过利用目标矢量和码矢量来获得误差标准，并通过利用所确定的搜索范围内的误差标准来获得最佳码矢量。

公开(公告)号：US08744841B2

公开(公告)日：2014-06-03

申请号：US11524274

申请日：2006-09-21

Applicant: Eun Mi Oh ,  Ki Hyun Choo ,  Jung-Hoe Kim ,  Chang Yong Son

Inventor： Eun Mi Oh ,  Ki Hyun Choo ,  Jung-Hoe Kim ,  Chang Yong Son

IPC: G10L19/00

CPC classification number: G10L19/0208 ,  G10L19/22

Abstract: An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.

Abstract translation: 一种自适应时间/频率编码模式确定装置，包括：时域特征提取单元，用于通过分析输入音频信号的时域信号来产生时域特征;频域特征提取单元，用于产生相应的频域特征; 通过对输入音频信号的频域信号的分析和模式确定单元来确定通过将与输入音频信号的帧相对应的频域划分成多个频域而产生的每个频带，以确定任何一个 基于时间的编码模式和基于频率的编码模式，通过使用时域特征和频域特征，相对于每个频带。

公开(公告)号：US20140149125A1

公开(公告)日：2014-05-29

申请号：US13686015

申请日：2012-11-27

Applicant: Chang-yong SON ,  Eun-mi OH ,  Ki-hyun CHOO ,  Jung-hoe KIM

Inventor： Chang-yong SON ,  Eun-mi OH ,  Ki-hyun CHOO ,  Jung-hoe KIM

IPC: G10L21/00

CPC classification number: G10L21/038 ,  G10L19/0204 ,  G10L19/08 ,  G10L21/00

Abstract: Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.