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1. 发明公开

EP3564953A3 COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 有权转让
公开(公告)号：EP3564953A3
公开(公告)日：2020-02-26
申请号：EP19173050.6
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 , H04B1/66 , G10L25/45 , H03G7/00 , H04B1/64
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 substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuate segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

2. 发明公开

EP3564953A2 COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 有权转让
公开(公告)号：EP3564953A2
公开(公告)日：2019-11-06
申请号：EP19173050.6
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 , H04B1/66 , G10L25/45 , H03G7/00 , H04B1/64
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 substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuate segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

3. 发明公开

EP3176786A1 COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 有权转让
标题翻译：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。

4. 发明申请

WO2021032719A1 MULTI-LAG FORMAT FOR AUDIO CODING 审中-公开
公开(公告)号：WO2021032719A1
公开(公告)日：2021-02-25
申请号：PCT/EP2020/073067
申请日：2020-08-18
申请人： DOLBY INTERNATIONAL AB
发明人： VILLEMOES, Lars , LEHTONEN, Heidi-Maria , PURNHAGEN, Heiko , HEDELIN, Per
IPC分类号： G10L25/06 , G10L19/02 , G10L25/18 , G10L25/30
摘要： Described herein is a method of encoding an audio signal. The method comprises: generating a plurality of subband audio signals based on the audio signal; determining a spectral envelope of the audio signal; for each subband audio signal, determining autocorrelation information for the subband audio signal based on an autocorrelation function of the subband audio signal; and generating an encoded representation of the audio signal, the encoded representation comprising a representation of the spectral envelope of the audio signal and a representation of the autocorrelation information for the plurality of subband audio signals. Further described are methods of decoding the audio signal from the encoded representation, as well as corresponding encoders, decoders, computer programs, and computer-readable recording media.

5. 发明申请

WO2010136459A1 EFFICIENT COMBINED HARMONIC TRANSPOSITION 审中-公开
标题翻译：有效的组合谐波传输
公开(公告)号：WO2010136459A1
公开(公告)日：2010-12-02
申请号：PCT/EP2010/057176
申请日：2010-05-25
申请人： DOLBY INTERNATIONAL AB , EKSTRAND, Per , VILLEMOES, Lars , HEDELIN, Per
发明人： EKSTRAND, Per , VILLEMOES, Lars , HEDELIN, Per
IPC分类号： G10L21/02 , G10L21/04
CPC分类号： G10H1/0091 , G10H1/125 , G10H2210/311 , G10L19/265 , G10L21/038 , G10L21/0388
摘要： The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular; a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described, The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δ f , The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P ; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P ; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of F Δ f ; with F being a resolution factor, with F ≥1; wherein the transposition order P is different from the resolution factor F .
摘要翻译：本文件涉及利用用于高频重构（HFR）的谐波转置方法的音频编码系统以及数字效果处理器，例如，所谓的兴奋剂，其中谐波失真的产生增加处理信号的亮度。尤其是; 描述了被配置为从信号的低频分量产生信号的高频分量的系统。系统可以包括分析滤波器组（501），其被配置为提供来自所述信号的低频分量的一组分析子带信号信号; 其中所述一组分析子带信号包括至少两个分析子带信号; 其中所述分析滤波器组（501）具有Δf的频率分辨率。所述系统还包括非线性处理单元（502），其被配置为使用转置顺序P从所述一组分析子带信号中确定一组合成子带信号; 其中所述合成子带信号集合包括所述一组分析子带信号的一部分，所述分析子带信号的相位偏移量是从所述置换顺序P导出的量; 以及合成滤波器组（504），被配置为从所述合成子带信号的集合生成所述信号的高频分量; 其中所述合成滤波器组（504）具有F ff的频率分辨率; F为解析因子，F = 1; 其中转置次数P与分辨率因子F不同

6. 发明申请

WO2022078960A1 SIGNAL CODING USING A GENERATIVE MODEL AND LATENT DOMAIN QUANTIZATION 审中-公开
公开(公告)号：WO2022078960A1
公开(公告)日：2022-04-21
申请号：PCT/EP2021/078053
申请日：2021-10-11
申请人： DOLBY INTERNATIONAL AB
发明人： KLEJSA, Janusz , VILLEMOES, Lars , HEDELIN, Per
IPC分类号： G10L19/02 , G06N3/04 , G06N3/0454
摘要： The present disclosure provides a decoder configured to receive a finite bitrate stream that includes a quantized latent frame, where the quantized latent frame includes a quantized representation of a current frame of a signal in a latent domain different from a first domain; to generate a reconstructed latent frame from the quantized latent frame; to use a generative neural network model to perform a task for which the general neural network model has been trained, wherein the task includes to generate parameters for an invertible mapping from the latent domain to the first domain; to reconstruct a current frame of the signal in the first domain, which includes to map the reconstructed latent frame to the first domain by use of the invertible mapping, and to use the reconstructed current frame of the signal in the first domain to update a state of the generative neural network model.

7. 发明申请

WO2014161994A2 ADVANCED QUANTIZER 审中-公开
标题翻译：先进的量化器
公开(公告)号：WO2014161994A2
公开(公告)日：2014-10-09
申请号：PCT/EP2014/056855
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 un-dithered quantizers (323). The quantization unit (112) is further configured to determine an SNRindication 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）; 和一个或多个非抖动量化器（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指示从量化器集合（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。

8. 发明申请

WO2010081892A3 CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION 审中-公开
公开(公告)号：WO2010081892A3
公开(公告)日：2010-07-22
申请号：PCT/EP2010/050483
申请日：2010-01-15
申请人： DOLBY INTERNATIONAL AB , VILLEMOES, Lars , HEDELIN, Per
发明人： VILLEMOES, Lars , HEDELIN, Per
IPC分类号： G10L21/02
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.

9. 发明申请

WO2020089215A1 METHODS AND APPARATUS FOR RATE QUALITY SCALABLE CODING WITH GENERATIVE MODELS 审中-公开
公开(公告)号：WO2020089215A1
公开(公告)日：2020-05-07
申请号：PCT/EP2019/079508
申请日：2019-10-29
申请人： DOLBY INTERNATIONAL AB
发明人： KLEJSA, Janusz , HEDELIN, Per
IPC分类号： G10L19/24 , G10L25/30
摘要： Described herein is a method of decoding an audio or speech signal, the method including the steps of: (a) receiving, by a decoder, a coded bitstream including the audio or speech signal and conditioning information; (b) providing, by a bitstream decoder, decoded conditioning information in a format associated with a first bitrate; (c) converting, by a converter, the decoded conditioning information from the format associated with the first bitrate to a format associated with a second bitrate; and (d) providing, by a generative neural network, a reconstruction of the audio or speech signal according to a probabilistic model conditioned by the conditioning information in the format associated with the second bitrate. Described are further an apparatus for decoding an audio or speech signal, a respective encoder, a system of the encoder and the apparatus for decoding an audio or speech signal as well as a respective computer program product.

10. 发明申请

WO2014165543A1 COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 审中-公开
标题翻译：使用高级光谱扩展降低量化噪声的装置和方法
公开(公告)号：WO2014165543A1
公开(公告)日：2014-10-09
申请号：PCT/US2014/032578
申请日：2014-04-01
申请人： DOLBY LABORATORIES LICENSING CORPORATION , DOLBY INTERNATIONAL AB
发明人： HEDELIN, Per , BISWAS, Arijit , SCHUG, Michael , MELKOTE, Vinay
IPC分类号： G10L21/034 , H03G3/24 , G03G7/00 , 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 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：实施例涉及用于减少音频编解码器中的编码噪声的压扩方法和系统。压缩处理通过压缩处理来降低初始音频信号的原始动态范围，该压缩处理使用定义的窗口形状将初始音频信号分成多个段，使用基于非能量的平均值来计算频域中的宽带增益初始音频信号的频域样本，并且应用各个增益值来放大相对较低强度的片段并衰减相对较高强度的片段。压缩的音频信号然后被扩展回到基本上原始的动态范围，该动态范围应用反向增益值来放大相对较高强度的片段，并且衰减相对较低强度的段。使用QMF滤波器组来分析初始音频信号以获得频域表示。

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