公开(公告)号：KR100480341B1

公开(公告)日：2005-03-31

申请号：KR1020030015683

申请日：2003-03-13

Applicant: 한국전자통신연구원

Inventor： 성호상 ,  황대환

IPC: G10L19/04

CPC classification number: G10L19/24

Abstract: 본 발명은 음성신호 처리에 관한 것으로 특히 광대역 음성 신호의 부호화기에 관한 것이며 보다 상세하게는 광대역 저전송률 음성 신호의 부호화기에 관한 것으로, 고주파수로 샘플링된 음성 신호 프레임을 저주파수로 다운 샘플링 한 후, DC성분이 제거된 음성 신호 프레임을 생성하는 전처리 및 다운샘플링 블록; 상기 다운 샘플링된 음성 신호 프레임을 입력받아 상기 프레임의 선형 예측 계수를 구하고 이 계수를 ISP로 변환하여 양자화 시키며 상기 ISP의 인덱스를 생성하는 LPC분석 및 ISP양자화 블록; 상기 다운 샘플링된 음성 신호의 합성 필터의 여기 신호를 모델링하기 위한 잔차 신호를 산출하는 잔차신호 계산 블록; 상기 여기 신호의 모델링을 위한 랜덤벡터를 발생시키는 랜덤벡터 발생 블록; 상기 랜덤벡터를 스케일링하기 위한 이득을 산출하는 이득계산 블록; 및 상기 이득을 양자화하고, 상기 이득의 인덱스를 생성하는 이득 양자화 블록을 포함한다.

公开(公告)号：KR100465316B1

公开(公告)日：2005-01-13

申请号：KR1020020071497

申请日：2002-11-18

Applicant: 한국전자통신연구원

Inventor： 성호상

IPC: G10L19/12 ,  G10L19/107

Abstract: PURPOSE: A vocoder and a voice coding method using the vocoder are provided to reduce the quantity of calculations required for fixed codebook search for innovation analysis and decrease the number of memories and bits used for the fixed codebook search. CONSTITUTION: A vocoder includes a pre-processor for pre-processing an input voice signal, a linear predictive coding analysis and quantization unit for carrying out linear predictive coding analysis and quantization for the pre-processed signal, a pitch analyzer for analyzing a pitch to obtain long-section correlation, and a codebook searching unit for searching an algebraic codebook to model an innovation signal. The vocoder further includes a synthesis filter for synthesizing a speech signal using the pitch and code vector searched by the pitch analyzer and the codebook searching unit and a linear predictive coding coefficient output from the linear predictive coding analysis and quantization unit, an adder for calculating an error between the outputs of the pre-processor and the synthesis filter, a weight filter for receiving the error obtained by the adder to output a weight error, and a parameter encoder for encoding the linear predictive coding coefficient, a pitch parameter selected by the pitch analyzer and a codebook parameter selected by the codebook searching unit to output encoded voice data.

Abstract translation: 目的：提供一种使用声码器的声码器和声码编码方法，以减少用于固定码本搜索所需的计算量以进行创新分析，并减少用于固定码本搜索的存储器和比特的数量。 一种声码器，包括用于预处理输入语音信号的预处理器，用于对预处理信号执行线性预测编码分析和量化的线性预测编码分析和量化单元，用于分析音调的音调分析器， 获得长期相关性;以及码本搜索单元，用于搜索代数码本以对创新信号建模。 声码器还包括：合成滤波器，用于使用由音调分析器和码本搜索单元搜索的音调和编码矢量以及从线性预测编码分析和量化单元输出的线性预测编码系数来合成语音信号;加法器，用于计算 预处理器和合成滤波器的输出之间的误差，接收由加法器获得的误差以输出加权误差的加权滤波器，以及用于对线性预测编码系数进行编码的参数编码器，由音调选择的音调参数 分析器和由码簿搜索单元选择的码簿参数，以输出经编码的语音数据。

公开(公告)号：KR1020040078760A

公开(公告)日：2004-09-13

申请号：KR1020030013606

申请日：2003-03-05

Applicant: 한국전자통신연구원

Inventor： 성호상 ,  황대환 ,  강상원 ,  이강은

IPC: G10L19/07 ,  G10L19/038

CPC classification number: G10L19/07

Abstract: PURPOSE: An LSF(Line Spectral Frequency) coefficient vector quantizer for broadband voice coding is provided to reduce memory capacity and the quantity of calculations required for quantization and prevent the deterioration of the performance of the quantizer. CONSTITUTION: An LSF coefficient vector quantizer includes a prediction quantizer(30), an non-prediction quantizer(31), and a switch(32). The prediction quantizer includes the first vector quantizer(VQ1) for non-structurally quantizing an LSF coefficient vector to produce a candidate vector to be quantized, a predictor for calculating a predicted LSF vector of the LSF coefficient vector, and the first lattice quantizer for lattice-quantizing the candidate vector with reference to the predicted LSF vector to produce a final prediction quantization vector of the LSF coefficient vector. The non-prediction quantizer includes the second vector quantizer(VQ2) for non-structurally quantizing the LSF coefficient vector to produce a candidate vector to be quantized, and the second lattice quantizer for lattice-quantizing the candidate vector to produce a final non-prediction quantization vector of the LSF coefficient vector. The switch decides one of the final prediction quantization vector and the final non-prediction quantization vector, which has smaller difference from the LSF coefficient vector as a final quantization vector of the LSF coefficient vector.

Abstract translation: 目的：提供用于宽带语音编码的LSF（线谱频率）系数矢量量化器，以减少存储容量和量化所需的计算量，并防止量化器性能的恶化。 构成：LSF系数矢量量化器包括预测量化器（30），非预测量化器（31）和开关（32）。 预测量化器包括用于非结构量化LSF系数向量以产生要量化的候选矢量的第一矢量量化器（VQ1），用于计算LSF系数矢量的预测LSF矢量的预测器和用于晶格的第一晶格量化器 - 参考预测的LSF向量量化候选向量，以产生LSF系数向量的最终预测量化矢量。 非预测量化器包括用于非结构量化LSF系数向量以产生要量化的候选向量的第二矢量量化器（VQ2），以及用于对候选矢量进行晶格量化以产生最终非预测的第二格子量化器 LSF系数向量的量化向量。 开关决定作为LSF系数向量的最终量化矢量的与LSF系数矢量的差较小的最终预测量化矢量和最终的非预测量化矢量之一。

公开(公告)号：KR100396844B1

公开(公告)日：2003-09-02

申请号：KR1020000080996

申请日：2000-12-23

Applicant: 한국전자통신연구원

Inventor： 박찬원 ,  이종형 ,  황대환 ,  성호상

IPC: H04L12/66

Abstract: PURPOSE: An Internet phone system and a method for operating the same are provided to use various codecs, to eliminate a voice delay with the optimum configuration and control of a voice buffer, to achieve home networking, to generate various tones, to minimize network loads, and to embody a rapid and stable Internet phone. CONSTITUTION: An Internet phone system is comprised of a home PNA transceiver part(111), an ethernet transceiver part(112), a MAC(Media Access Controller) part(109), a G.729/G.723 software codec part(102), a VoIP_IF part(105), and a G.711 codec part(113). The home PNA transceiver part(111) covers home networking in a physical layer. The ethernet transceiver part(112) covers ethernet connection in the physical layer. The MAC part(109) takes charge of access control with the Internet. The G.729/G.723 software codec part(102) executes a compression/decompression function for voice data inputted through the home PNA transceiver part(111) or the ethernet transceiver part(112). The VoIP_IF part(105) covers a function to process the voice data compressed or decompressed by the G.729/G.723 software codec part(102). The G.711 codec part(113) executes A/D conversion, D/A conversion or A/y-law PCM conversion for tones or voice data.

Abstract translation: 目的：提供一种因特网电话系统及其操作方法，以使用各种编解码器，通过对语音缓冲器的最佳配置和控制来消除语音延迟，实现家庭联网，生成各种音调，以最小化网络负载 ，并体现快速稳定的网络电话。 构成：因特网电话系统包括家庭PNA收发器部分（111），以太网收发器部分（112），MAC（媒体访问控制器）部分（109），G.729 / G.723软件编解码器部分 102），VoIP_IF部分（105）和G.711编解码器部分（113）。 家庭PNA收发器部分（111）覆盖物理层中的家庭网络。 以太网收发器部分（112）覆盖物理层中的以太网连接。 MAC部分（109）负责与因特网的访问控制。 G.729 / G.723软件编解码器部分（102）对通过家庭PNA收发器部分（111）或以太网收发器部分（112）输入的语音数据执行压缩/解压缩功能。 VoIP_IF部分（105）覆盖处理由G.729 / G.723软件编解码器部分（102）压缩或解压缩的语音数据的功能。 G.711编解码器部分（113）对音调或语音数据执行A / D转换，D / A转换或A / Y律PCM转换。

公开(公告)号：KR100388488B1

公开(公告)日：2003-06-25

申请号：KR1020000083271

申请日：2000-12-27

Applicant: 한국전자통신연구원

Inventor： 성호상

IPC: G11B27/022

Abstract: PURPOSE: A high-speed pitch searching method in a voiced sound section is provided to perform search only around a pitch of a prior frame, if a frame decided as a voiced sound is continued in a voice coder, so as to reduce calculation through the variation of the pitch. CONSTITUTION: A voice codec of a voice coder is initialized(201). If a voice signal of one frame is inputted(202), VAD(Voice Activity Detection) by the frame is performed(203). If the VAD is completed, information on the activity of the current frame and information on a pitch of a prior frame are used to decide a pitch search mode(204,205). The voice signal is coded using the decided pitch search mode(206), and the voice activity and pitch data of the current frame are stored by the coded information(207). Whether the current voice signal is a final signal is decided(208). If so, the steps are ended, and if not, the step of 202 is returned.

Abstract translation: 目的：如果在语音编码器中判定为有声声音的帧持续下去，则在有声声音部分中提供高速音调搜索方法以仅在先前帧的音调周围执行搜索，以便减少通过 球场的变化。 构成：话音编码器的话音编解码器被初始化（201）。 如果输入了一帧的语音信号（202），则执行该帧的VAD（语音活动检测）（203）。 如果VAD完成，则使用关于当前帧的活动的信息和关于先前帧的音调的信息来确定音调搜索模式（204,205）。 语音信号使用确定的音调搜索模式（206）被编码，并且当前帧的语音活动和音调数据由编码信息（207）存储。 决定当前语音信号是否是最终信号（208）。 如果是，则结束步骤，否则返回步骤202。

公开(公告)号：KR100494555B1

公开(公告)日：2005-06-10

申请号：KR1020010081121

申请日：2001-12-19

Applicant: 한국전자통신연구원

Inventor： 성호상 ,  황대환

IPC: H04M11/06

CPC classification number: G10L19/18 ,  G10L2021/02082

Abstract: 본 발명은 음성신호를 전송하는 방법 및 그 장치에 관한 것으로, 본 발명에 따른 음성신호 전송장치는 음성신호를 입력받아 고대역 신호와 저대역 신호로 분리하여 출력하는 직교반사필터부, 상기 저대역 신호를 입력받아 부호화하는 저대역부호화부, 상기 고대역 신호를 입력받아 부호화하는 고대역부호화부 및 음성신호가 전송될 통신망의 종류에 따라서 상기 부호화된 고대역 신호 및 저대역 신호를 다중화하여 상기 통신망으로 전송하는 통신망 접속부를 포함하는 것을 특징으로 한다.
본 발명에 따르면 제공되는 통신망의 종류에 따라서 다양하게 음성신호를 다중화 시켜 전달하게 되므로 효율적으로 음성신호를 전송할 수 있게되는 효과가 있다.

公开(公告)号：KR100467326B1

公开(公告)日：2005-01-24

申请号：KR1020020077996

申请日：2002-12-09

Applicant: 한국전자통신연구원 ,  학교법인연세대학교

Inventor： 성호상 ,  황대환 ,  윤대희 ,  강홍구 ,  박영철 ,  이기승 ,  정성교 ,  김경태

IPC: G10L19/002 ,  G10L19/02 ,  G10L19/04

CPC classification number: G10L19/24 ,  G10L21/038

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract translation: 本发明涉及通过使用附加比特分配方法进行语音编码和解码的发射机和接收机。 根据本发明的发射机和接收机通过使用系统要求中允许的附加比特来实现高质量的语音通信服务，同时使用常规的语音编码器。 另外，根据本发明的发射机和接收机具有这样的优点，即它们能够插入额外的量化块，而不改变常规标准语音编码器的结构，因为它们通过应用多级量化过程来分配附加比特 不在语音信号域中，而是在参数域中。

公开(公告)号：KR1020050006883A

公开(公告)日：2005-01-17

申请号：KR1020030046861

申请日：2003-07-10

Applicant: 한국전자통신연구원 ,  학교법인연세대학교

Inventor： 성호상 ,  황대환 ,  김경태 ,  정성교 ,  강홍구 ,  윤대희

IPC: G10L19/12

CPC classification number: G10L19/125

Abstract: PURPOSE: A wideband voice encoder, a method therefor, a wideband decoder and a method therefor are provided to offer excellent voice quality in a voice interval which is not processed by an algebraical codebook by performing multi-stage fixed codebook retrieval. CONSTITUTION: A voice characteristic classifying unit(105) classifies the characteristic of a voice corresponding to a current frame using an open-loop pitch value of a recognition weight filtered signal of a wideband voice signal and a linear prediction factor by a statistical method. An adaptive codebook retrieving unit(106) retrieves a pitch delay value near the open-loop pitch value, calculates a pitch gain value, and generates an adaptive codebook contribution signal. The adaptive codebook retrieving unit outputs difference between the generated adaptive codebook contribution signal and the recognition weight filtered signal as a primary fixed codebook target signal. A primary fixed codebook retrieving unit(107) obtains a primary fixed codebook index and a primary fixed codebook gain value, generates a primary fixed codebook contribution signal corresponding to the obtained primary fixed codebook index. The primary fixed codebook retrieving unit outputs different between the generated primary fixed codebook contribution signal and the primary fixed codebook target signal as a secondary codebook target signal. A secondary fixed codebook retrieving unit(108) has at least two or more fixed codebooks according to voice characteristics, selects one secondary fixed codebook according to voice characteristic information, and retrieves secondary fixed codebook indexes and secondary fixed codebook gain values according to the voice characteristics. A parameter multiplexing unit(110) multiplexes voice characteristic information, the pitch delay value, the pitch gain value, the primary fixed codebook index, the primary fixed codebook gain value, the secondary fixed codebook indexes and the secondary fixed codebook gain values, generates a bit stream, and transmits the bit stream to an external voice decoding terminal.

Abstract translation: 目的：提供一种宽带语音编码器，其方法，宽带解码器及其方法，以通过执行多级固定码本检索在代码码本不处理的语音间隔中提供优良的语音质量。 构成：语音特征分类单元（105）使用统计方法使用宽带语音信号的识别权重滤波信号和线性预测因子的开环音调值来分类与当前帧相对应的语音的特性。 自适应码本检索单元（106）检索开环音调值附近的音调延迟值，计算音调增益值，并生成自适应码本贡献信号。 自适应码本检索单元输出生成的自适应码本贡献信号和识别权重滤波信号之间的差作为主要固定码本目标信号。 主固定码本检索单元（107）获得主固定码本索引和主固定码本增益值，生成与所获得的主固定码本索引相对应的主固定码本贡献信号。 主固定码本检索单元在所生成的主固定码本贡献信号和主固定码本目标信号之间输出不同的第二码本目标信号。 第二固定码本检索单元（108）根据语音特征具有至少两个或多个固定码本，根据语音特征信息选择一个辅助固定码本，并根据语音特征检索辅助固定码本索引和辅助固定码本增益值 。 参数复用单元（110）多路复用语音特征信息，音调延迟值，音调增益值，主固定码本索引，主固定码本增益值，辅助固定码本索引和辅助固定码本增益值，生成 并将比特流发送到外部语音解码终端。

公开(公告)号：KR1020040055073A

公开(公告)日：2004-06-26

申请号：KR1020020081663

申请日：2002-12-20

Applicant: 한국전자통신연구원 ,  학교법인연세대학교

Inventor： 성호상 ,  황대환 ,  윤대희 ,  강홍구 ,  박영철 ,  이기승 ,  정성교 ,  김경태

IPC: H04M1/00 ,  H04B1/50

CPC classification number: H04L12/66

Abstract: PURPOSE: An apparatus and a method for transmitting/receiving a wideband voice signal are provided to offer high sound quality 16 kHz sampled wideband voice signal and compatibility with an existing system. CONSTITUTION: A analyzing/filtering unit(301) receives a digital voice signal and outputs a low band voice signal having a bandwidth of 0-4 kHz of 8 kHz sampling period. A low band standard coding unit(302) receives the low band voice signal, codes it and outputs a coded low band standard signal. A low band additional coding unit(304) receives a difference between a filtered low band voice signal and a composite signal that has been decoded by a low band standard decoding unit(303) and codes them to generate a low band additional signal. An enhancement residual band coding unit(306) up-samples a composite signal obtained by combining signals outputted from the low band standard decoding unit(303) and the low band additional decoding unit(305), obtains a difference between the up-sampled signal and a voice signal inputted to the analyzing/filtering unit(301), and generates an enhancement residual band signal.

Abstract translation: 目的：提供用于发送/接收宽带语音信号的装置和方法，以提供高音质16kHz采样宽带语音信号和与现有系统的兼容性。 构成：分析/滤波单元（301）接收数字语音信号并输出​​具有8kHz采样周期的0-4kHz带宽的低频语音信号。 低频标准编码单元（302）接收低频语音信号，对其编码并输出编码的低频带标准信号。 低频带附加编码单元（304）接收滤波后的低频语音信号和由低频带标准解码单元（303）解码的复合信号之间的差，并对它们进行编码以产生低频带附加信号。 增强残差频带编码单元（306）对通过组合从低频带标准解码单元（303）和低频带附加解码单元（305）输出的信号获得的复合信号进行上采样，获得上采样信号 以及输入到分析/滤波单元（301）的语音信号，并产生增强剩余频带信号。

公开(公告)号：KR100651712B1

公开(公告)日：2006-11-30

申请号：KR1020030046861

申请日：2003-07-10

Applicant: 한국전자통신연구원 ,  학교법인연세대학교

Inventor： 성호상 ,  황대환 ,  김경태 ,  정성교 ,  강홍구 ,  윤대희

IPC: G10L19/12

CPC classification number: G10L19/125

Abstract: 광대역 음성 부호화기 및 그 방법과 광대역 음성 복호화기 및 그 방법이 개시된다. 본 발명에 따른 광대역 음성 부호화기는 부호화할 광대역 음성신호의 개회로 피치값과 선형예측계수를 이용하여 현재 프레임에 해당하는 음성의 특성을 규정하는 음성 특성 분류부, 적응 코드북을 검색하여 적응 코드북 피치 지연값 및 적응 코드북 피치 이득값을 얻고, 1차 고정 코드북 목적신호를 생성하는 적응 코드북 검색부, 1차 고정 코드북을 검색하여 1차 고정 코드북 인덱스와 1차 고정 코드북 이득값을 얻고, 2차 고정 코드북 목적신호를 생성하는 1차 고정 코드북 검색부, 음성 특성에 따라 적어도 둘 이상의 2차 고정 코드북들을 구비하며, 음성 특성 정보에 따라 하나의 2차 고정 코드북을 선택 및 검색하여 2차 고정 코드북 인덱스들과 2차 고정 코드북 이득값들 검색하는 2차 고정코드북 검색부 및 각 부에서 얻어지는 파라미터들을 양자화 및 다중화하여 비트열로 만들어 외부의 음성 복호화단으로 전송하는 파라미터 다중화부를 포함하는 것을 특징으로 하며, 음성 특성에 따라 2개 이상으로 구성된 2차 고정 코드북들로부터 음성 특성에 적합한 2차 고정 코드북을 선택함으로써 광대역 음성신호에 대해 보다 우수한 음질을 제공할 수 있다.