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    METHODS AND ARRANGEMENTS FOR LOUDNESS AND SHARPNESS COMPENSATION IN AUDIO CODECS
    1.
    发明公开
    METHODS AND ARRANGEMENTS FOR LOUDNESS AND SHARPNESS COMPENSATION IN AUDIO CODECS 有权
    方法和安排：音频编解码器数量和聚焦补偿

    公开(公告)号：EP2502229A4

    公开(公告)日：2013-06-19

    申请号：EP10831864

    申请日：2010-06-29

    Applicant: ERICSSON TELEFON AB L M

    Inventor： GRANCHAROV VOLODYA ,  SVERRISSON SIGURDUR

    IPC: G10L21/038 ,  G10L19/26

    CPC classification number: G10L21/038 ,  G10L19/265

    Abstract: In a method of improving perceived loudness and sharpness of a reconstructed speech signal delimited by a predetermined bandwidth, performing the steps of providing (S10) the speech signal, and separating (S20) the provided signal into at least a first and a second signal portion. Subsequently, adapting (S30) the first signal portion to emphasize at least a predetermined frequency or frequency interval within the first bandwidth portion. Finally, reconstructing (S40) the second signal portion based on at least the first signal portion, and combining (S50) the adapted first signal portion and the reconstructed second signal portion to provide a reconstructed speech signal with an overall improved perceived loudness and sharpness.

    METHOD AND ARRANGEMENT FOR PROCESSING OF AUDIO SIGNALS
    2.
    发明公开
    METHOD AND ARRANGEMENT FOR PROCESSING OF AUDIO SIGNALS 有权
    方法和系统用于处理音频信号

    公开(公告)号：EP2569767A4

    公开(公告)日：2013-10-02

    申请号：EP11780883

    申请日：2011-04-28

    Applicant: ERICSSON TELEFON AB L M

    Inventor： GRANCHAROV VOLODYA ,  SVERRISSON SIGURDUR

    IPC: G10L19/26

    CPC classification number: G10L19/26 ,  G10L19/02

    Abstract: Method and decoder for processing of audio signals. The method and decoder relate to deriving a processed vector {circumflex over (d)} by applying a post-filter directly on a vector d comprising quantized MDCT domain coefficients of a time segment of an audio signal. The post-filter is configured to have a transfer function H which is a compressed version of the envelope of the vector d. A signal waveform is reconstructed by performing an inverse MDCT transform on the processed vector {circumflex over (d)}.

    FILING OF NON-CODED SUB-VECTORS IN TRANSFORM CODED AUDIO SIGNALS
    6.
    发明公开
    FILING OF NON-CODED SUB-VECTORS IN TRANSFORM CODED AUDIO SIGNALS 有权
    文件中未编码的次载体的转化CODED音频信号

    公开(公告)号：EP2684190A4

    公开(公告)日：2014-08-13

    申请号：EP11860593

    申请日：2011-09-14

    Applicant: ERICSSON TELEFON AB L M

    Inventor： GRANCHAROV VOLODYA ,  NÄSLUND SEBASTIAN ,  SVERRISSON SIGURDUR

    IPC: G10L19/028 ,  G10L19/00 ,  G10L19/02

    CPC classification number: G10L19/02 ,  G10L19/0212 ,  G10L19/028 ,  G10L19/038 ,  G10L21/038 ,  G10L2019/0007

    Abstract: An apparatus for filling non-coded residual sub-vectors of a transform coded audio signal. The apparatus comprises means for compressing coded residual sub-vectors, means for rejecting compressed residual sub-vectors that do not fulfill a predetermined criterion and means for concatenating the remaining compressed residual sub-vectors to form a first virtual codebook. The apparatus further comprises means for combining pairs of coefficients of the first virtual codebook to form a second virtual codebook, means for filling non-coded residual sub-vectors below a predetermined frequency with coefficients from the first virtual codebook and means for filling non-coded residual sub-vectors above the predetermined frequency with coefficients from the second virtual codebook.

    IMPROVED EXCITATION SIGNAL BANDWIDTH EXTENSION
    7.
    发明公开
    IMPROVED EXCITATION SIGNAL BANDWIDTH EXTENSION 有权
    带宽扩展FOR改进激励信号

    公开(公告)号：EP2502230A4

    公开(公告)日：2013-05-15

    申请号：EP10831865

    申请日：2010-07-05

    Applicant: ERICSSON TELEFON AB L M

    Inventor： SVERRISSON SIGURDUR ,  BRUHN STEFAN ,  GRANCHAROV VOLODYA

    IPC: G10L21/02 ,  G10L19/04

    CPC classification number: G10L19/12 ,  G10L21/038

    Abstract: An apparatus for generating a high band extension of a low band excitation signal (eLB) defined by parameters representing a CELP encoded audio signal includes the following elements: upsamplers (20) configured to upsample a low band fixed codebook vector (uFCB) and a low band adaptive codebook vector (uACB) to a predetermined sampling frequency. A frequency shift estimator (22) configured to determine a modulation frequency (&OHgr;) from an estimated measure representing a fundamental frequency (F0) of the audio signal. A modulator (24) configured to modulate the upsampled low band adaptive codebook vector (uACB↑) with the determined modulation frequency to form a frequency shifted adaptive codebook vector. A compression factor estimator (28) configured to estimate a compression factor. A compressor (34) configured to attenuate the frequency shifted adaptive codebook vector and the upsampled fixed codebook vector (uFCB↑) based on the estimated compression factor. A combiner (40) configured to form a high-pass filtered sum of the attenuated frequency shifted adaptive codebook vector and the attenuated up-sampled fixed codebook vector.

    METHOD AND ARRANGEMENT FOR PROCESSING OF AUDIO SIGNALS
    8.
    发明申请
    METHOD AND ARRANGEMENT FOR PROCESSING OF AUDIO SIGNALS 审中-公开
    用于处理音频信号的方法和装置

    公开(公告)号：WO2011142709A3

    公开(公告)日：2011-12-29

    申请号：PCT/SE2011050518

    申请日：2011-04-28

    Applicant: ERICSSON TELEFON AB L M ,  GRANCHAROV VOLODYA ,  SVERRISSON SIGURDUR

    Inventor： GRANCHAROV VOLODYA ,  SVERRISSON SIGURDUR

    IPC: G10L19/14

    CPC classification number: G10L19/26 ,  G10L19/02

    Abstract: Method and decoder for processing of audio signals. The method and decoder relate to deriving a processed vector d by applying a post-filter directly on a vector d comprising quantized MDCT domain coefficients of a time segment of an audio signal. The post-filter is configured to have a transfer function H which is a compressed version of the envelope of the vector d. A signal wave form is reconstructed by performing an inverse MDCT transform on the processed vector d.

    Abstract translation: 用于处理音频信号的方法和解码器。 该方法和解码器涉及通过在包括音频信号的时间段的量化的MDCT域系数的矢量d上直接应用后置滤波器来导出处理后的矢量d。 后置滤波器被配置为具有传递函数H，其是向量d的包络的压缩版本。 通过对处理的向量d执行逆MDCT变换来重建信号波形。

    AUDIO ENCODING/DECODING BASED ON AN EFFICIENT REPRESENTATION OF AUTO-REGRESSIVE COEFFICIENTS
    9.
    发明申请
    AUDIO ENCODING/DECODING BASED ON AN EFFICIENT REPRESENTATION OF AUTO-REGRESSIVE COEFFICIENTS 审中-公开
    基于自回归系数的有效表示的音频编码/解码

    公开(公告)号：WO2013066236A2

    公开(公告)日：2013-05-10

    申请号：PCT/SE2012050520

    申请日：2012-05-15

    Applicant: ERICSSON TELEFON AB L M ,  GRANCHAROV VOLODYA ,  SVERRISSON SIGURDUR

    Inventor： GRANCHAROV VOLODYA ,  SVERRISSON SIGURDUR

    IPC: G10L19/06

    CPC classification number: G10L19/038 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/06 ,  G10L21/038 ,  G10L2019/001

    Abstract: Described is an encoder (50) for encoding a parametric spectral representation (f) of auto-regressive coefficients that partially represent an audio signal. The encoder includes a low-frequency encoder (10) configured to quantize elements of a part of the parametric spectral representation that correspond to a low-frequency part of the audio signal. It also includes a high-frequency encoder (12) configured to encode a high-frequency part (f H ) of the parametric spectral representation (f) by weighted averaging based on the quantized elements (f L ) flipped around a quantized mirroring frequency (f m ), which separates the low-frequency part from the high- frequency part, and a frequency grid determined from a frequency grid codebook (24) in a closed-loop search procedure. Described are also a corresponding decoder, corresponding encoding/decoding methods and UEs including such an encoder/decoder.

    Abstract translation: 描述了用于对部分表示音频信号的自回归系数的参数频谱表示（f）进行编码的编码器（50）。 编码器包括配置成量化与音频信号的低频部分对应的参数频谱表示的一部分的元素的低频编码器（10）。 它还包括高频编码器（12），该高频编码器被配置为通过基于在量化的镜像频率（f L）周围翻转的量化元素（f L）通过加权平均来编码参数频谱表示（f）的高频部分（f H） fm），其将低频部分与高频部分分开，并且在闭环搜索过程中从频率网格码本（24）确定频率网格。 还描述了相应的解码器，对应的编码/解码方法和包括这种编码器/解码器的UE。

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