-
公开(公告)号：EP1187337A4
公开(公告)日：2005-05-11
申请号：EP99913723
申请日：1999-04-19
申请人： FUJITSU LTD
发明人： SUZUKI MASANAO , OTA YASUJI , TSUCHINAGA YOSHITERU
摘要： Optimal speech coding is conducted in response to inputted speech. In an adaptation code book (Ba) are stored signal vector sequences of past speech signals. Vector extracting means (11) extracts a signal vector and neighborhood vectors stored near the signal vector. A high-order long-time prediction synthetic filter (12) generates a long-time prediction speech signal (Sna-1) from the signal vector and the neighborhood signal vectors. Filter coefficient calculating means (13) calculates the filter coefficient of the long-time prediction synthetic filter (12). An audition weighting synthetic filter (14) generates a reproduction coded speech signal (Sna) from the long-time prediction speech signal (Sna-1). Error calculating means (15) calculates the error (En) between the speech signal (Sn) and the reproduction coded speech signal (Sna). Minimum error detecting means (16) detects the minimum error from among the calculated errors. Optimal value transmitting means (17) transmits the optimal filter coefficient ( beta a) and the optimal delay (La) when the minimum value is detected as optimal values.
-
公开(公告)号：EP1775718A4
公开(公告)日：2008-05-07
申请号：EP04770880
申请日：2004-07-22
申请人： FUJITSU LTD
CPC分类号： G10L19/025 , G10L19/0212 , G10L19/022
摘要： An audio encoding apparatus and an audio encoding method, wherein the processing amount can be reduced and a block length can be appropriately selected. An electric power calculation part (402) calculates, from an input signal, an electric power variation ratio. A predicted gain variation ratio calculation part (406) calculates, from the input signal, a predicted gain variation ratio. A block length determination part (407) determines, from the electric power variation ratio and the predicted gain variation ratio, whether an encoding using a long block is to be executed or an encoding using a short block is to be executed. Based on this determination, a long block MDCT conversion part (409) or a short block MDCT conversion part (410) performs a discrete cosine transformation of the input signal.
-
3.发明公开AUDIO PACKET VANISHMENT CONCEALING DEVICE, AUDIO PACKET VANISHMENT CONCEALING METHOD, RECEPTION TERMINAL, AND AUDIO COMMUNICATION SYSTEM 有权装置和方法用于隐藏音频包的消失，和EMPFANGSENDGERûTAUDIOKOMMUNIKAITONSSYSTEM
公开(公告)号：EP1589330A4
公开(公告)日：2007-07-11
申请号：EP03815584
申请日：2003-01-30
申请人： FUJITSU LTD
CPC分类号： G10L19/005 , H04L29/06027 , H04L65/607 , H04L65/80 , H04M7/006
-
公开(公告)号：EP2768242A4
公开(公告)日：2015-04-29
申请号：EP11873877
申请日：2011-10-14
申请人： FUJITSU LTD
发明人： ENDO KAORI , TSUCHINAGA YOSHITERU
CPC分类号： H04R3/002 , G10L21/0208 , G10L2021/02082 , G10L2021/02166 , H04R3/005 , H04R3/02 , H04R2499/11
摘要： [Object] To provide high-quality sound while suppressing the calculation amount [Solution Means] A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.
-
公开(公告)号：EP1221694A4
公开(公告)日：2005-06-22
申请号：EP99943314
申请日：1999-09-14
申请人： FUJITSU LTD
发明人： SUZUKI MASANAO , OTA YASUJI , TSUCHINAGA YOSHITERU
CPC分类号： G10L19/09 , G10L19/04 , G10L19/107 , G10L19/18
摘要： Disclosed is a voice encoding method having a synthesis filter implemented using linear prediction coefficients obtained by dividing an input signal into frames each of a fixed length, and subjecting the input signal to linear prediction analysis in the frame units, generating a reconstructed signal by driving said synthesis filter by a periodicity signal output from an adaptive codebook and a pulsed signal output from an algebraic codebook, and performing encoding in such a manner that an error between the input signal and said reproduced signal is minimized, wherein there are provided an encoding mode 1 that uses pitch lag obtained from an input signal of a present frame and an encoding mode 2 that uses pitch lag obtained from an input signal of a past frame. Encoding is performed in encoding mode 1 and encoding mode 2, the mode in which the input signal can be encoded more precisely is decided frame by frame and encoding is carried out on the basis of the mode decided.
-
6.发明公开TWO-WAY COMMUNICATION SYSTEM, COMMUNICATION INSTRUMEN T, AND COMMUNICATION CONTROL METHOD 有权双向通信系统，通信仪器以及通信控制方法
公开(公告)号：EP1542422A4
公开(公告)日：2010-12-29
申请号：EP02777960
申请日：2002-10-25
申请人： FUJITSU LTD
CPC分类号： H04L1/0014 , H04L1/0021 , H04M3/24 , H04M7/006 , H04M7/0072
-
公开(公告)号：EP1557827A4
公开(公告)日：2008-05-14
申请号：EP02779956
申请日：2002-10-31
申请人： FUJITSU LTD
CPC分类号： G10L19/06 , G10L21/0364
摘要： A voice intensifier capable of reducing abrupt changes in the amplification factor between frames and realizing excellent sound quality with less noise feeling by dividing input voices into the sound source characteristic and the vocal tract characteristic, so as to individually intensify the sound source characteristic and the vocal tract characteristic and then synthesize them before being output. The voice intensifier comprises a signal separation unit for separating the input sound signal into the sound source characteristic and the vocal tract characteristic, a characteristic extraction unit for extracting characteristic information from the vocal tract characteristic, a corrective vocal tract characteristic calculation unit for obtaining vocal tract characteristic correction information from the vocal tract characteristic and the characteristic information, a vocal tract characteristic correction unit for correcting the vocal tract characteristic by using the vocal tract characteristic correction information, and a signal synthesizing means for synthesizing the corrective vocal tract characteristic from the vocal tract characteristic correction unit and the sound source characteristic, so that the sound synthesized by the signal synthesizing means is output.
摘要翻译： 一种语音增强器，能够减少帧间放大因子的突然变化，并通过将输入语音划分为声源特性和声道特性来实现具有较小噪声感的优良音质，从而单独增强声源特性和声音 然后在输出之前合成它们。 声音增强器包括：信号分离单元，用于将输入声音信号分离为声源特性和声道特性;特征提取单元，用于从声道特征提取特征信息;修正声道特征计算单元，用于获取声道 来自声道特性和特征信息的特征性校正信息;声道特性校正单元，用于通过使用声道特性校正信息来校正声道特性;以及信号合成装置，用于合成来自声道的校正声道特性 特性校正单元和声源特性，从而输出由信号合成装置合成的声音。
-
公开(公告)号：EP2161720A4
公开(公告)日：2012-06-13
申请号：EP07767257
申请日：2007-06-20
申请人： FUJITSU LTD
IPC分类号： G10L19/02 , G10L19/00 , G10L19/032 , H03M7/30
CPC分类号： G10L19/032 , G10L19/005
摘要： Disclosed is a decoding apparatus for decoding corded data obtained by encoding each of a scale value and a spectrum value of frequency domain audio signal data to output an audio signal. The decoding apparatus includes a unit configured to decode and inversely quantize the coded data to obtain the frequency domain audio signal data, a unit configured to compute from the coded data one of the number of scale bits composed of the number of bits corresponding to the scale value of the coded data and the number of spectrum bits composed of the number of bits corresponding to the spectrum value of the coded data, a unit configured to estimate a quantization error of the frequency domain audio signal data based on one of the number of scale bits and the number of spectrum bits of the coded data, a unit configured to compute a correction amount based on the estimated quantization error and correct the frequency domain audio signal data obtained by the frequency domain data obtaining unit based on the computed correction amount, and a unit configured to convert the corrected frequency domain audio signal data into the audio signal.
-
公开(公告)号：EP1933305A4
公开(公告)日：2009-08-26
申请号：EP05776793
申请日：2005-09-05
申请人： FUJITSU LTD
IPC分类号： G10L19/02
CPC分类号： G10L19/035 , G10L25/27
摘要： It is possible to reduce audio quality degradation caused by a pre-echo and bit shortage. An acoustic analysis unit (11) analyzes an audio signal and acquires a perception entropy as a parameter expressing the number of bits required for quantization. An encoding bit quantity monitoring unit (12) monitors the number of encoded bits when an audio signal is encoded and acquires an excessive number of bits as the number of bits which can be used in the current frame. According to a combination of the perception entropy and the excessive number of bits, a frame division quantity decision unit (13) decides the division quantity for dividing the one frame of the audio signal into N from 1 to N. An orthogonal conversion unit (14) divides the one frame by the decided division quantity and performs orthogonal conversion of the audio signal by the divided block length unit so as to obtain an orthogonal conversion coefficient. A quantization unit (15) quantizes the orthogonal conversion coefficient by the block length unit.
-
公开(公告)号：EP1619666A4
公开(公告)日：2007-08-01
申请号：EP03721013
申请日：2003-05-01
申请人： FUJITSU LTD
CPC分类号： G10L19/26 , G10L19/12 , G10L21/0364 , G10L25/15
-
-
-
-
-
-
-
-
-
搜索结果
10 条记录 (耗时 0.008 秒)
