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13 条记录 (耗时 0.022 秒)

    CONCEPT FOR COMBINED DYNAMIC RANGE COMPRESSION AND GUIDED CLIPPING PREVENTION FOR AUDIO DEVICES
    4.
    发明公开
    CONCEPT FOR COMBINED DYNAMIC RANGE COMPRESSION AND GUIDED CLIPPING PREVENTION FOR AUDIO DEVICES 审中-公开

    公开(公告)号：EP3951778A1

    公开(公告)日：2022-02-09

    申请号：EP21186145.5

    申请日：2014-10-20

    申请人： FRAUNHOFER-GESELLSCHAFT zur Förderung der angewandten Forschung e.V.

    发明人： KUECH, Fabian ,  UHLE, Christian ,  KRATSCHMER, Michael ,  NEUGEBAUER, Bernhard ,  MEIER, Michael ,  BORSUM, Arne

    IPC分类号： G10L21/034 ,  H03G9/00 ,  H03G11/00 ,  G10L19/16 ,  G10L19/008

    摘要： The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept comprises
    an audio processing chain configured to receive a decoded audio signal derived from the audio bitstream and to adjust characteristics of the audio signal in order to produce an audio output signal, the audio adjustment chain comprising a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and
    a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

    APPARATUS FOR MERGING SPATIAL AUDIO STREAMS
    5.
    发明公开
    APPARATUS FOR MERGING SPATIAL AUDIO STREAMS 有权
    设备公司合并空间音频流

    公开(公告)号：EP2324645A1

    公开(公告)日：2011-05-25

    申请号：EP09806392.8

    申请日：2009-08-11

    申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

    发明人： DEL GALDO, Giovanni ,  KUECH, Fabian ,  KALLINGER, Markus ,  PULKKI, Ville ,  LAITINEN, Mikko-Ville ,  SCHULTZ-AMLING, Richard

    IPC分类号： H04S3/00

    CPC分类号： H04S3/008 ,  G10L19/008 ,  H04S2420/03 ,  H04S2420/11

    摘要： An apparatus (100) for merging a first spatial audio stream with a second spatial audio stream to obtain a merged audio stream comprising an estimator (120) for estimating a first wave representation comprising a first wave direction measure and a first wave field measure for the first spatial audio stream, the first spatial audio stream having a first audio representation and a first direction of arrival. The estimator (120) being adapted for estimating a second wave representation comprising a second wave direction measure and a second wave field measure for the second spatial audio stream, the second spatial audio stream having a second audio representation and a second direction of arrival. The apparatus (100) further comprising a processor (130) for processing the first wave representation and the second wave representation to obtain a merged wave representation comprising a merged wave field measure and a merged direction of arrival measure, and for processing the first audio representation and the second audio representation to obtain a merged audio representation, and for providing the merged audio stream comprising the merged audio representation and the merged direction of arrival measure.

    ECHO SUPPRESSION COMPRISING MODELING OF LATE REVERBERATION COMPONENTS
    10.
    发明公开
    ECHO SUPPRESSION COMPRISING MODELING OF LATE REVERBERATION COMPONENTS 有权
    与模拟后期混响成分的回声抑制

    公开(公告)号：EP2633522A1

    公开(公告)日：2013-09-04

    申请号：EP11767430.9

    申请日：2011-10-06

    申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

    发明人： KUECH, Fabian ,  SCHMIDT, Markus ,  FALLER, Christof ,  FAVROT, Alexis

    IPC分类号： G10L21/02 ,  H04M9/08

    CPC分类号： H04R3/002 ,  G10L2021/02082 ,  H04M9/085

    摘要： An apparatus (200) for computing filter coefficients (H[k,m]) for an adaptive filter (210) is disclosed. The adaptive filter is used for filtering a microphone signal (140, 430) so as to suppress an echo due to a loudspeaker signal (130, 420). The apparatus (200) comprises: an echo decay modeling means (465) for modeling a decay behavior of an acoustic environment (120) and for providing a corresponding echo decay parameter (τ; α
    m ); and computing means (270; 370) for computing the filter coefficients (H[k,m]) of the adaptive filter (210) on the basis of the echo decay parameter (τ; α
    m ). A corresponding method comprises: providing echo decay parameters (τ; Ĝ[k,m]; α
    m ) determined by means of an echo decay modeling means (465); and computing the filter coefficients (H[k,m]) of the adaptive filter (210) on the basis of the echo decay parameters (τ; α
    m ).