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    COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION
    3.
    发明公开
    COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 审中-公开
    扩和方法来降低量化晚期频谱扩张

    公开(公告)号：EP3176786A1

    公开(公告)日：2017-06-07

    申请号：EP16205100.7

    申请日：2014-04-01

    申请人： Dolby Laboratories Licensing Corp. ,  Dolby International AB

    发明人： HEDELIN, Per ,  BISWAS, Arijit ,  SCHUG, Michael ,  MELKOTE, Vinay

    IPC分类号： G10L21/034 ,  G10L25/18 ,  H03G3/24 ,  G10L25/45 ,  H03G7/00 ,  H04B1/64

    CPC分类号： G10L21/0232 ,  G10L19/008 ,  G10L19/0208 ,  G10L19/032 ,  G10L21/034 ,  G10L25/18 ,  G10L25/45 ,  H03G7/007 ,  H04B1/66

    摘要： Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

    摘要翻译： 实施例涉及用于在音频编解码器中减少编码噪声压扩方法和系统。 的通过一个压缩processthat分割初始音频信号划分成使用所定义的windowshape段的多元化的初始音频信号的原始动态范围压缩处理减少，计算使用的基于非能量平均在频域中的宽带增益 初始音频信号的频域样本，并应用个别的增益值来放大相对低强度的段和衰减相对高强度的段。 然后，将压缩的音频信号扩展回基本上与原始动态范围没有适用逆增益值来放大相对高强度的段和相对低强度的衰减段。 甲QMF滤波器组来分析初始音频信号以获得频域表示。

    CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION
    4.
    发明公开
    CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION 审中-公开

    公开(公告)号：EP4300495A3

    公开(公告)日：2024-02-21

    申请号：EP23210729.2

    申请日：2010-01-15

    申请人： Dolby International AB

    发明人： VILLEMOES, Lars ,  HEDELIN, Per

    IPC分类号： G10L21/0388 ,  G10L25/90

    摘要： The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

    CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION
    5.
    发明公开
    CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION 审中-公开

    公开(公告)号：EP3992966A1

    公开(公告)日：2022-05-04

    申请号：EP21209274.6

    申请日：2010-01-15

    申请人： Dolby International AB

    发明人： VILLEMOES, Lars ,  HEDELIN, Per

    IPC分类号： G10L21/0388 ,  G10L25/90

    摘要： The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

    AUDIO ENCODER AND DECODER
    6.
    发明公开
    AUDIO ENCODER AND DECODER 审中-公开

    公开(公告)号：EP3671738A1

    公开(公告)日：2020-06-24

    申请号：EP19200800.1

    申请日：2014-04-04

    申请人： Dolby International AB

    发明人： VILLEMOES, Lars ,  KLEJSA, Janusz ,  HEDELIN, Per

    IPC分类号： G10L19/02 ,  G10L19/06 ,  G10L19/032

    摘要： The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a transform-based audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder (100, 170) configured to encode a speech signal into a bitstream is described. The encoder (100, 170) comprises a framing unit (101) configured to receive a set (132, 332) of blocks; wherein the set (132, 332) of blocks comprises a plurality of sequential blocks (131) of transform coefficients; wherein the plurality of blocks (131) is indicative of samples of the speech signal; wherein a block (131) of transform coefficients comprises a plurality of transform coefficients for a corresponding plurality of frequency bins (301). Furthermore, the encoder (100, 170) comprises an envelope estimation unit (102) configured to determine a current envelope (133) based on the plurality of sequential blocks (131) of transform coefficients; wherein the current envelope (133) is indicative of a plurality of spectral energy values (303) for the corresponding plurality of frequency bins (301). In addition, the encoder (100, 170) comprises an envelope interpolation unit (104) configured to determine a plurality of interpolated envelopes (136) for the plurality of blocks (131) of transform coefficients, respectively, based on the current envelope (133); Furthermore, the encoder (100, 170) comprises a flattening unit (108) configured to determine a plurality of blocks (140) of flattened transform coefficients by flattening the corresponding plurality of blocks (131) of transform coefficients using the corresponding plurality of interpolated envelopes (136), respectively; wherein the bitstream is determined based on the plurality of blocks (140) of flattened transform coefficients.

    ADVANCED QUANTIZER
    9.
    发明公开
    ADVANCED QUANTIZER 审中-公开
    先进的量化器

    公开(公告)号：EP3217398A1

    公开(公告)日：2017-09-13

    申请号：EP17164112.9

    申请日：2014-04-04

    申请人： Dolby International AB

    发明人： KLEJSA, Janusz ,  VILLEMOES, Lars ,  HEDELIN, Per

    IPC分类号： G10L19/035

    CPC分类号： G10L19/035 ,  G10L19/005 ,  G10L19/028 ,  G10L19/20

    摘要： The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a transform-based audio codec system which is particularly well suited for voice encoding/decoding. A quantization unit (112) configured to quantize a first coefficient of a block (141) of coefficients is described. The block (141) of coefficients comprises a plurality of coefficients for a plurality of corresponding frequency bins (301). The quantization unit (112) is configured to provide a set (326, 327) of quantizers. The set (326, 327) of quantizers comprises a plurality of different quantizers (321, 322, 323) associated with a plurality of different signal-to-noise ratios, referred to as SNR, respectively. The plurality of different quantizers (321, 322, 323) includes a noise-filling quantizer (321); one or more dithered quantizers (322); and one or more undithered quantizers (323). The quantization unit (112) is further configured to determine an SNR indication indicative of a SNR attributed to the first coefficient, and to select a first quantizer from the set (326, 327) of quantizers, based on the SNR indication. In addition, the quantization unit (112) is configured to quantize the first coefficient using the first quantizer.

    摘要翻译： 本文件涉及音频编码和解码系统（称为音频编解码器系统）。 特别地，本文件涉及特别适合于语音编码/解码的基于变换的音频编解码器系统。 描述被配置为量化系数块（141）的第一系数的量化单元（112）。 系数块（141）包括多个对应频率仓（301）的多个系数。 量化单元（112）被配置为提供量化器的集合（326,327）。 量化器组（326,327）包括分别与多个不同的信噪比（称为SNR）相关联的多个不同的量化器（321,322,323）。 多个不同的量化器（321,322,323）包括噪声填充量化器（321） 一个或多个抖动量化器（322）; 和一个或多个未交替的量化器（323）。 量化单元（112）还被配置为确定指示归因于第一系数的SNR的SNR指示，并且基于SNR指示从量化器集（326,327）中选择第一量化器。 另外，量化单元（112）被配置为使用第一量化器量化第一系数。