公开(公告)号：EP2116037A1

公开(公告)日：2009-11-11

申请号：EP07859382.9

申请日：2007-12-13

申请人： Nokia Corporation

发明人： VIROLAINEN, Jussi ,  LAAKSONEN, Laura ,  AHMANIEMI, Ali ,  VALVE, Päivi

IPC分类号： H04M3/56 ,  H04M1/725

CPC分类号： H04M3/56 ,  H04M1/7253 ,  H04M3/562 ,  H04M3/564 ,  H04M3/568 ,  H04M2250/06 ,  H04M2250/62 ,  H04W4/00 ,  H04W8/26 ,  H04W48/08

摘要： Provided are multichannel architectures, systems, methods, and computer program products for distributed teleconferencing using one or more master devices and/or a centralized conferencing switch. Multichannels enhance functionality of a master device in distributed teleconferencing and allow for compatibility with 3D capable teleconferencing. Multichannel distributed teleconferencing involves multichannel, monophonic, and/or a fixed number of uplink and downlink channels. A multichannel distributed teleconferencing system may perform active talker detection of near-end participants and communicate an ID signal on an uplink channel identifying the active near-end participants. A multichannel distributed teleconferencing system may also receive an ID signal on a downlink channel identifying the active far-end participants. A multichannel distributed teleconferencing system may perform various uplink and downlink processing. Uplink processing may involve multimixing and spatialization. Multimixing may be used to separate speech signals of near-end participants. Spatialization, also used in downlink processing, introduces spatial separation of active participants.

公开(公告)号：EP1581929A2

公开(公告)日：2005-10-05

申请号：EP04701060.8

申请日：2004-01-09

申请人： Nokia Corporation

发明人： KALLIO, Laura ,  ALKU, Paavo ,  KÄYHKÖ, Kimmo ,  KAJALA, Matti ,  VALVE, Päivi

IPC分类号： G10L19/06

CPC分类号： G10L21/038 ,  G10L25/93

摘要： A method and device for improving the quality of speech signals transmitted using an audio bandwidth between 300 Hz and 3.4 kHz. After the received speech signal is divided into frames, zeros are inserted between samples to double the sampling frequency. The level of these aliased frequency components is adjusted using an adaptive algorithm based on the classification of the speech frame. Sound can be classified into sibilants and non-sibilants, and a non-sibilant sound can be further classified into a voiced sound and a stop consonant. The adjustment is based on parameters, such as the number of zero-crossings and energy distribution, computed from the spectrum of the up-sampled speech signal between 300 Hz and 3.4kHz. A new sound with a bandwidth between 300 Hz and 7.7kHz is obtained by inverse Fourier transforming the spectrum of the adjusted, up-sampled sound.