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公开(公告)号：KR100554164B1
公开(公告)日：2006-02-22
申请号：KR1020030047455
申请日：2003-07-11
CPC classification number: G10L19/173 , G10L19/12
Abstract: 서로 다른 CELP 방식의 음성 코덱 간의 상호부호화 장치 및 그 방법이 개시된다. 본 발명에 따른 서로 다른 CELP 방식의 음성 코덱 간의 상호부호화 장치는 서로 다른 포맷을 갖는 입력 CELP 코덱과 출력 CELP 코덱 사이에서, 입력 CELP 코덱의 포맷을 출력 CELP 코덱의 포맷으로 변환하는 본 발명에 따른 상호부호화 장치는 입력 CELP 코덱 포맷으로 부호화된 비트스트림를 음성 신호로 변환하는 입력 CELP 코덱의 복호화 처리부, 기준필터를 기준으로, 스펙트럼 왜곡이 최소가 되는 최적의 가중치를 적용하여 계산되는 필터 특성으로 입력 CELP 코덱의 복호화 처리부에서 복호화된 음성신호를 필터 처리하는 상호부호화 필터, 다수의 가중치로 이루어진 가중치 세트로부터 상호부호화 필터의 스펙트럼 왜곡을 최소화하는 최적의 가중치를 추출하여 상호부호화 필터로 제공하는 상호부호화 필터 설계부 및 상호부호화 필터에서 필터 처리된 음성신호를 부호화하여 출력 CELP 코덱 포맷의 비트스트림를 생성하는 출력 CELP 코덱의 부호화 처리부를 포함하는 것을 특징으로 하고, 하나의 상호부호화 필터를 이용하여 종래의 후-필터 및 지각가중필터를 대신함으로써, 상호부호화기의 연산량을 감소시키면서도 수신단에서 복호화된 음성의 음질을 향상시킬 수 있다.
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公开(公告)号：KR100550003B1
公开(公告)日：2006-02-08
申请号：KR1020030047454
申请日：2003-07-11
Abstract: 상호부호화기에서 개회로 피치 추정 방법 및 그 장치가 개시된다. 서로 다른 CELP 방식의 음성 코덱 간의 상호부호화를 위한 상호부호화기에서 본 발명에 따른 개회로 피치 추정 방법은, 입력 CELP 코덱 포맷으로 부호화된 비트스트림를 음성 신호로 복호화하고, 복호화된 음성신호의 각 부-프레임에 대한 폐-루프 피치를 추출하는 (a)단계, 복호화된 음성 신호를 사람의 청각기관 특성을 고려한 지각가중필터 처리를 하는 (b)단계, 복호화된 폐-루프 피치와, 이전 프레임의 마지막 부-프레임에 대한 폐-루프 피치 또는 동일 프레임에 대한 이전 개-루프 피치를 이용하여 결정된 개-루프 피치 검색 범위에서 지각가중필터링된 음성신호의 개-루프 피치를 검색하는 (c)단계 및 검색된 개-루프 피치를 이용하여 결정된 폐-루프 피치 검색 범위에서 지각가중 필터링된 음성신호의 폐-루프 피치를 검색하고, 검색된 결과를 출력 CELP 코덱 포맷의 피치 지연값으로서 생성하는 (d)단계를 포함하는 것을 특징으로 하며, 기존 재-검색 방식보다 음질 저하를 줄일 수 있으며, 계산량 감소 효과를 얻을 수 있다.
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公开(公告)号：KR1020040050811A
公开(公告)日：2004-06-17
申请号：KR1020020077996
申请日：2002-12-09
IPC: G10L19/002 , G10L19/02 , G10L19/04
CPC classification number: G10L19/24 , G10L21/038
Abstract: PURPOSE: A transceiver for encoding and decoding voice using an additional bit assignment method is provided to offer a high-quality voice service by assigning only additional bits admitted in a system using a voice encoder used in an old voice processing system. CONSTITUTION: A standard voice encoding unit(102) classifies an inputted voice signal into spectrum information indicating a vocal tract transfer function and an excitation signal component, and models and quantizes spectrum information and the excitation signal component. The standard voice encoding unit(102) performs the standard encoding of the quantized signal, and generates a standard-encoded bit series. A quality improvement encoding unit(103) obtains an error between a signal in which spectrum information is not quantized and a signal in which spectrum information is quantized by the standard voice encoding unit(102). The quality improvement encoding unit(103) obtains an error between a signal in which the excitation signal component is not quantized and a signal in which the excitation signal component is quantized. The quality improvement encoding unit(103) additionally quantizes the obtained errors, and generates an encoded bit series. A multiplexing unit(104) multiplexes the bit series obtained in the standard voice encoding unit(102) and the quality improvement encoding unit(103), and transmits the multiplexed bit series.
Abstract translation: 目的：提供一种用于使用附加位分配方法对语音进行编码和解码的收发器，以通过仅使用在旧语音处理系统中使用的语音编码器仅分配系统中允许的附加比特来提供高质量语音服务。 构成：标准语音编码单元（102）将输入的语音信号分类为指示声道传递函数和激励信号分量的频谱信息，并对频谱信息和激励信号分量进行建模和量化。 标准语音编码单元（102）执行量化信号的标准编码，并生成标准编码比特序列。 质量改进编码单元（103）获得频谱信息未被量化的信号与频谱信息由标准语音编码单元（102）量化的信号之间的误差。 质量改进编码单元（103）获得其中激励信号分量未被量化的信号与激励信号分量被量化的信号之间的误差。 质量改进编码单元（103）另外量化所获得的误差，并生成编码比特序列。 复用单元（104）复用在标准语音编码单元（102）和质量改进编码单元（103）中获得的比特序列，并且发送多路复用的比特序列。
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Patent Agency Ranking
公开(公告)号：KR1020050048214A
公开(公告)日：2005-05-24
申请号：KR1020030082086
申请日：2003-11-19
Applicant: 학교법인연세대학교
Abstract: 본 발명은 화자인식시스템의 화자 특징벡터 생성방법 및 시스템에 관한 것으로서, 입력된 화자 음성신호에서 특징벡터를 추출하여 화자모델링 훈련 및 화자인식을 수행하는 화자인식시스템에 있어서, 입력된 화자음성신호의 피치간격을 측정하고 소정의 피치구간 음성신호를 추출하는 피치구간 음성추출수단, 피치구간음성추출수단에서 추출된 음성신호의 특징벡터를 생성하는 특징벡터생성수단을 포함하여 이루어진 것을 특징으로 한다.
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公开(公告)号：KR100467326B1
公开(公告)日：2005-01-24
申请号：KR1020020077996
申请日：2002-12-09
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。
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公开(公告)号：KR1020050007854A
公开(公告)日：2005-01-21
申请号：KR1020030047455
申请日：2003-07-11
CPC classification number: G10L19/173 , G10L19/12
Abstract: PURPOSE: An apparatus and a method for transcoding data between speech codecs having different CELP(Code Excited Linear Prediction) types are provided to reduce quantity of calculation and offer speech with high quality by utilizing one transcoding filter instead of a post-processing filter and a perception weighting filter. CONSTITUTION: A decoding process part(321) converts a bitstream with an input CELP codec format into a speech signal, wherein the bitstream is formed by coding an input speech signal. A transcoding filter(323) filters the decoded speech signal by filter characteristics calculated by adopting an optimal weighting value for minimum spectrum distortion. A transcoding filter design part(322) extracts the optimal weighting value from a weighting value set comprised of a plurality of weighting value pairs. A coding process part(324) codes the speech signal processed by the transcoding filter and thereby generates a bitstream of an output CELP codec format.
Abstract translation: 目的：提供一种用于在具有不同CELP（代码激励线性预测）类型的语音编解码器之间对数据进行代码转换的装置和方法，以通过利用一个代码转换滤波器而不是后处理滤波器来减少计算量并提供高质量的语音， 感知加权滤波器 构成：解码处理部分（321）将具有输入CELP编解码器格式的比特流转换为语音信号，其中通过对输入的语音信号进行编码来形成比特流。 代码转换滤波器（323）通过针对最小频谱失真采用最佳加权值计算的滤波特性来对解码的语音信号进行滤波。 代码转换滤波器设计部分（322）从由多个加权值对组成的加权值集合中提取最优加权值。 编码处理部分（324）对由代码转换滤波器处理的语音信号进行编码，从而生成输出CELP编解码器格式的比特流。
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17.发明公开
公开(公告)号：KR1020050006883A
公开(公告)日：2005-01-17
申请号：KR1020030046861
申请日：2003-07-10
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。
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公开(公告)号：KR1020040055073A
公开(公告)日：2004-06-26
申请号：KR1020020081663
申请日：2002-12-20
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。
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