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.
Abstract:
A system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, encoding a difference between a baseband speech signal and a standard baseband between a synthesized standard baseband signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.
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:
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:
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:
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:
PURPOSE: A stereo matching apparatus and method are provided to output a disparity map having divided objects and background by performing stereo matching through the preprocessing of an inputted image. CONSTITUTION: A stereo matching method is as follows. A stereo matching apparatus receives binocular images obtained from first and second cameras (S100). The stereo matching apparatus increases the contradiction between an object and a background in the obtained images using a preset first algorithm (S200). The stereo matching apparatus performs stereo matching using the processed images (S300). [Reference numerals] (AA) Start; (BB) End; (S100) Input a binocular image; (S200) Image processing through a first algorithm; (S300) Perform stereo matching
Abstract:
얼굴 검출 장치 및 이를 이용한 거리 측정 방법에서는, 스테레오 카메라로부터 획득된 좌우 영상을 이용하여 얼굴이 검출된다. 동시에 스테레오 정합 과정 없이 오직 스테레오 카메라로부터 제공된 영상 프레임을 이용하여 스테레오 카메라로부터 얼굴까지의 거리가 측정된다. 따라서, 저성능의 시스템에서도, 얼굴 검출 기능과 거리 측정 기능이 동시 수행될 수 있는 저성능 시스템의 설계가 가능하다.
Abstract:
얼굴 검출 시스템 및 그 방법에서, 전처리 영상에 대응하는 전처리 계수 값들이 병렬적으로 산출되는 윈도우 추출부와, 병렬적으로 산출된 상기 전처리 계수 값들에 근거하여 코스트값을 병렬적으로 산출하는 코스트값 계산부가 구비된다. 이러한 병렬 연산 기법에 의해 전체 시스템의 연산처리속도가 향상되고, 그 결과, 고속의 실시간으로 얼굴 패턴의 검출이 가능하다.
Abstract:
본 발명은 개개의 화자식별 결과의 신뢰도 측정 방법에 관한 것으로, 각 프레임의 화자식별 결과에 대한 공헌 정도를 측정하고, 이 각 프레임의 화자식별 공헌도를 기반으로 화자식별 결과의 신뢰도를 측정하고, 이를 화자 진위 판단에 이용함으로써, 화자 검증시에 제시된 화자의 진위를 정확하게 판단하여, 원거리 다채널 환경에서 화자식별의 정확도를 높일 수 있다. 화자식별, 화자식별 결과의 신뢰도, 원거리 다채널 환경, 화자식별 결과 통합.