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    ENABLING SAMPLING RATE DIVERSITY IN A VOICE COMMUNICATION SYSTEM
    2.
    发明申请
    ENABLING SAMPLING RATE DIVERSITY IN A VOICE COMMUNICATION SYSTEM 审中-公开

    公开(公告)号：US20200082836A1

    公开(公告)日：2020-03-12

    申请号：US16685016

    申请日：2019-11-15

    Applicant: DOLBY INTERNATIONAL AB

    Inventor： Heiko PURNHAGEN ,  Leif SEHLSTROM ,  Lars VILLEMOES ,  Glenn N. DICKINS ,  Mark S. VINTON

    IPC: G10L19/03 ,  H04M3/56 ,  G10L19/16 ,  G10L19/002 ,  G10L19/26

    Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

    AUDIO DECODER FOR INTERLEAVING SIGNALS
    5.
    发明申请
    AUDIO DECODER FOR INTERLEAVING SIGNALS 审中-公开

    公开(公告)号：US20200098381A1

    公开(公告)日：2020-03-26

    申请号：US16593830

    申请日：2019-10-04

    Applicant: DOLBY INTERNATIONAL AB

    Inventor： Kristofer KJOERLING ,  Heiko PURNHAGEN ,  Harald MUNDT ,  Karl Jonas ROEDEN ,  Leif SEHLSTROM

    IPC: G10L19/20 ,  G10L19/008 ,  G10L25/18 ,  H04S3/00 ,  G10L19/02 ,  G10L19/16

    Abstract: A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal comprising spectral coefficients corresponding to frequencies up to a first cross-over frequency for a time frame and performing parametric decoding at a second cross-over frequency for the time frame to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method also includes extracting from the encoded audio bitstream a second waveform-coded signal comprising spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency for the time frame and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal for the time frame.

    AUDIO DECODING SYSTEM AND AUDIO ENCODING SYSTEM
    7.
    发明申请
    AUDIO DECODING SYSTEM AND AUDIO ENCODING SYSTEM 审中-公开
    音频解码系统和音频编码系统

    公开(公告)号：US20160232900A1

    公开(公告)日：2016-08-11

    申请号：US14915579

    申请日：2014-09-08

    Applicant: DOLBY INTERNATIONAL AB

    Inventor： Lars VILLEMOES ,  Leif Jonas SAMUELSSON ,  Kristofer KJOERLING ,  Heiko PURNHAGEN ,  Leif SEHLSTROM

    IPC: G10L19/008

    CPC classification number: G10L19/008

    Abstract: An audio decoding system (100) for processing a two-channel input signal (X) comprises a parametric mixing stage (110). The parametric mixing stage receives the two-channel input signal and a set of mixing parameters (P1), and outputs a two-channel output signal (Y1). The parametric mixing stage comprises a decorrelation stage (111) outputting a decorrelated signal (D1) based on the input signal. The parametric mixing stage further comprises a mixing matrix (112) receiving the input signal and the de-correlated signal, and forming a two-channel linear combination of channels from the input signal and the decorrelated signal. The mixing matrix outputs the linear combination as the two-channel output signal. Coefficients of the linear combination are controllable by the set of mixing parameters, and at least four mixing parameters of the set are independently assignable. In example embodiments, multiple parametric mixing stages are used to independently reconstruct additional channels encoded in the input signal.

    Abstract translation: 用于处理双通道输入信号（X）的音频解码系统（100）包括参数混合级（110）。 参数混合级接收双通道输入信号和一组混合参数（P1），并输出双通道输出信号（Y1）。 参数混合级包括基于输入信号输出解相关信号（D1）的去相关级（111）。 参数混合级还包括接收输入信号和去相关信号的混合矩阵（112），以及从输入信号和解相关信号形成通道的双通道线性组合。 混合矩阵输出线性组合作为双通道输出信号。 线性组合的系数可以通过混合参数的集合来控制，并且该组的至少四个混合参数是独立可分配的。 在示例实施例中，使用多个参数混合级来独立地重建在输入信号中编码的附加通道。

    AUDIO DECODER FOR INTERLEAVING SIGNALS
    9.
    发明申请
    AUDIO DECODER FOR INTERLEAVING SIGNALS 有权

    公开(公告)号：US20220059110A1

    公开(公告)日：2022-02-24

    申请号：US17463192

    申请日：2021-08-31

    Applicant: DOLBY INTERNATIONAL AB

    Inventor： Kristofer KJOERLING ,  Heiko PURNHAGEN ,  Harald MUNDT ,  Karl Jonas ROEDEN ,  Leif SEHLSTROM

    IPC: G10L19/20 ,  G10L19/008 ,  G10L25/18 ,  H04S3/00 ,  G10L19/02 ,  G10L19/16

    Abstract: A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal comprising spectral coefficients corresponding to frequencies up to a first cross-over frequency for a time frame and performing parametric decoding at a second cross-over frequency for the time frame to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method also includes extracting from the encoded audio bitstream a second waveform-coded signal comprising spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency for the time frame and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal for the time frame.

    Enabling Sampling Rate Diversity In A Voice Communication System
    10.
    发明申请
    Enabling Sampling Rate Diversity In A Voice Communication System 审中-公开

    公开(公告)号：US20180130479A1

    公开(公告)日：2018-05-10

    申请号：US15867318

    申请日：2018-01-10

    Applicant: DOLBY INTERNATIONAL AB

    Inventor： Heiko PURNHAGEN ,  Leif SEHLSTROM ,  Lars VILLEMOES ,  Glenn N. DICKINS ,  Mark S. VINTON

    IPC: G10L19/03 ,  G10L19/26 ,  H04M3/56 ,  G10L19/16 ,  G10L19/002

    CPC classification number: G10L19/03 ,  G10L19/002 ,  G10L19/16 ,  G10L19/26 ,  H04M3/56 ,  H04M3/568

    Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing moving spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

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