VORRICHTUNG UND VERFAHREN ZUM ANALYSIEREN EINES INFORMATIONSSIGNALS

    公开(公告)号:EP1743324B1

    公开(公告)日:2007-10-31

    申请号:EP05744658.5

    申请日:2005-04-29

    IPC分类号: G10L11/00

    CPC分类号: G10L25/48

    摘要: In order to analyse an information signal, a significant short-time spectrum is extracted from the information signal. The extraction device (16) is embodied in such a way as to extract the short-time spectra which come closer to a specific characteristic than other short-time spectra of the information signal. The extracted short-time spectra are then decomposed (18) into component signals, by ICA analysis, a component signal spectrum representing a profile spectrum of a sound source which generates a sound corresponding to the required characteristic. An amplitude envelope is calculated (20) for each profile spectrum from a series of short-time spectra of the information signal and from the determined profile spectra, said envelope indicating how the profile spectrum of a sound source generally varies over time. The profile spectra and associated amplitude envelopes describe the information signal that can be further evaluated, e.g. for the purposes of a transcription in the case of a music signal.

    VERFAHREN UND VORRICHTUNG ZUM EINBRINGEN VON INFORMATIONEN IN EINEN DATENSTROM SOWIE VERFAHREN UND VORRICHTUNG ZUM CODIEREN EINES AUDIOSIGNALS
    124.
    发明授权
    VERFAHREN UND VORRICHTUNG ZUM EINBRINGEN VON INFORMATIONEN IN EINEN DATENSTROM SOWIE VERFAHREN UND VORRICHTUNG ZUM CODIEREN EINES AUDIOSIGNALS 有权
    方法和设备中引入数据流的方法和设备信息以音频信号编码

    公开(公告)号:EP1212857B1

    公开(公告)日:2003-03-12

    申请号:EP00969432.4

    申请日:2000-10-05

    IPC分类号: H04H1/00 H04B1/66

    摘要: An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced. By the fact that the information to be introduced are introduced into the data stream without changing to the time domain, the block rastering underlying the short-term spectrum will not be touched, so that introducing a watermark will not lead to tandem encoding effects.

    VERFAHREN UND VORRICHTUNG ZUM CODIEREN VON AUDIOSIGNALEN SOWIE VERFAHREN UND VORRICHTUNGEN ZUM DECODIEREN EINES BITSTROMS
    126.
    发明授权
    VERFAHREN UND VORRICHTUNG ZUM CODIEREN VON AUDIOSIGNALEN SOWIE VERFAHREN UND VORRICHTUNGEN ZUM DECODIEREN EINES BITSTROMS 失效
    用于解码比特流的方法和设备,用于编码音频信号的方法和装置

    公开(公告)号:EP1025646B1

    公开(公告)日:2001-09-26

    申请号:EP98940163.3

    申请日:1998-07-07

    IPC分类号: H03M7/00

    CPC分类号: H04B1/665 H04B14/046

    摘要: The invention makes it possible to combine a scaleable audio coder with TNS technology. According to the inventive method for encoding time signals (x1) sampled in a first sampling rate, second time signals (x2) with a sampling rate smaller than the first sampling rate are generated (12). The second time signals (x2) are then encoded (14) according to a first coding algorithm, and written into a bit stream (xAUS) (16). The encoded second time signals (x2c) are then decoded (14) again and are transformed (23, 24) into the frequency range, as are the first time signals. TNS prediction coefficients are then calculated (25) from a spectral representation of the first time signals (X1). The transformed output signal (X2cd) of the coder/decoder (14) with the first coding algorithm and the spectral representation (X1) of the first time signal are subjected to a prediction of the frequency (27) in order to obtain spectral residual values for both signals using the prediction coefficients calculated on the basis of the first time signals alone. These two signals are evaluated against each other (26, 28). The evaluated spectral residual values (Xb) are then encoded by means of a second coding algorithm in order to obtain coded evaluated spectral residual values (Xcb). These evaluated spectral residual values are written into the bit stream (xAUS) in addition to side information with the prediction coefficients.

    VERFAHREN ZUM SIGNALISIEREN EINER RAUSCHSUBSTITUTION BEIM CODIEREN EINES AUDIOSIGNALS
    127.
    发明授权
    VERFAHREN ZUM SIGNALISIEREN EINER RAUSCHSUBSTITUTION BEIM CODIEREN EINES AUDIOSIGNALS 失效
    METHOD FOR信令的噪声替代在编码过程中的音频信号

    公开(公告)号:EP0931386B1

    公开(公告)日:2000-07-05

    申请号:EP98916947.9

    申请日:1998-03-13

    IPC分类号: H04B1/66

    CPC分类号: G10L19/028 H04B1/665

    摘要: The invention relates to a method for signalling a noise substitution during audio signal coding. According to said method, the audio signal is first transformed in the frequency range to obtain spectral values. The spectral values are subsequently grouped to form spectral value groups. On the basis of a detection whether a group of spectral values is a noise group or not, a coding table is allocated to a non-noise group or a tonal group by means of a coding table number for redundancy coding of the same. If a group is a noise group it is allocated an additional coding table number which does not refer to a coding table in order to signal that this group is a noise group and that it must not be redundancy coded. By signalling noise substitution by means of a Huffman-code table number for noise groups of spectral values which are for instance scale factor band sections and which must not be redundancy coded, an opportunity is provided for implementing availability of a noise substitution in a scale factor band in the bit flow syntax of the MPEG-2 Advanced Audio coding Standard, without intervening in the basic coding structure and without having to touch the structure of the existing bit flow syntax.