公开(公告)号：EP1700293A1

公开(公告)日：2006-09-13

申请号：EP04804095.0

申请日：2004-12-20

申请人： Fraunhofer-Gesellschaft Zur Förderung der angewand ten Forschung e.v.

发明人： SCHULLER, Gerald ,  LUTZKY, Manfred ,  KRÄMER, Ulrich ,  WABNIK, Stefan ,  HIRSCHFELD, Jens

IPC分类号： G10L19/14

CPC分类号： G10L19/04 ,  G10L19/025 ,  G10L19/24 ,  G10L2019/0014

摘要： The invention is based on an adaptive prediction algorithm which can be controlled by a speed coefficient in order to work at a first adaptation speed and with a first degree of precision of adaptation and degree of precision of prediction in the event that the speed coefficient has a first value and with a second degree of precision of adaptation, which is lower than the first, and with a degree of precision of prediction, which is higher than the first in the event that the speed parameter has a second value, wherein the periods of adaptation occurring after the reset times during which the prediction errors are initially increased as a result of the not yet adapted prediction coefficients, are reduced by initially adjusting the speed parameter to the first value (42) and after a while adjusting it to the second value (50). When the speed parameter is readjusted to the second value after a specific time period after the reset times, the prediction errors and the residuals that are to be transferred are optimized or smaller than with the first speed parameter value.

公开(公告)号：EP1745637A1

公开(公告)日：2007-01-24

申请号：EP05774310.6

申请日：2005-07-11

申请人： Fraunhofer-Gesellschaft Zur Förderung der angewand ten Forschung e.v.

发明人： TITTEL, Martin ,  LUTZKY, Manfred ,  HUPP, Jürgen

IPC分类号： H04M9/08

CPC分类号： H04M3/56 ,  H04M3/002 ,  H04M3/568 ,  H04M9/082

摘要： The invention relates to a conference terminal for a digital voice conferencing system. Said terminal comprises a first sound converter unit (56), which is configured to generate a microphone signal from an acoustic signal, a second sound converter unit (60), which is configured to generate an acoustic signal from a loudspeaker signal, a connecting device (64), which is configured to facilitate a connection between the conference terminal (50) and a central conference unit, in order to receive a collective conference signal from the central conference unit. The terminal also comprises a unit (52) for echo suppression, which is configured to combine the microphone signal or a signal derived from the latter in a listening mode with the collective conference signal in a noise-suppression device, to produce a loudspeaker signal containing a reduced acoustic signal, on which the microphone signal is based. A conference terminal of this type permits the suppression of a remote intrinsic echo and a remote extraneous echo. The invention permits the configuration of digital, wireless voice conferencing systems that guarantee a high voice-rendition quality and interference immunity with minimal wiring requirements.

公开(公告)号：EP1697928A1

公开(公告)日：2006-09-06

申请号：EP05707322.3

申请日：2005-02-10

申请人： Fraunhofer-Gesellschaft Zur Förderung der angewand ten Forschung e.v.

发明人： SCHULLER, Gerald ,  WABNIK, Stefan ,  HIRSCHFELD, Jens ,  LUTZKY, Manfred

IPC分类号： G10L19/00 ,  G10L19/14 ,  G10L19/02

CPC分类号： G10L19/02 ,  G10L19/0017 ,  G10L19/26 ,  G10L19/265

摘要： The aim of the invention is to encode an audio signal of a sequence of audio values into an encoded signal. To this end, a first monitoring threshold is determined for a first block of audio values of the sequence of audio values, and a second monitoring threshold is determined for a second block of audio values of the sequence of audio values; a version of a first parameterisation of a parameterisable filter is calculated such that the transmission function thereof corresponds approximately to the inverse of the quantity of the first monitoring threshold; a pre-determined block of audio values of the sequence of audio values is filtered by means of the parameterisable filter, using a pre-determined parameterisation that depends, in a pre-determined manner, on the version of the second parameterisation, in order to obtain a block of filtered audio values that corresponds to the pre-determined block; the filtered audio values are quantised in order to obtain a block of quantised filtered audio values; a combination of the version of the first parameterisation and the version of the second parameterisation, containing at least one difference between the version of the first parameterisation and the version of the second parameterisation, is formed; and information containing said combination and from which the quantified filtered audio values and a version of the first parameterisation can be derived is integrated into the encoded signal.