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1. 发明申请

US20060116874A1 Noise-dependent postfiltering 审中-公开
标题翻译：噪声依赖后置滤波
公开(公告)号：US20060116874A1
公开(公告)日：2006-06-01
申请号：US10540741
申请日：2003-10-24
申请人： Jonas Samuelsson , Willem Kleijn , Volodya Grancharov
发明人： Jonas Samuelsson , Willem Kleijn , Volodya Grancharov
IPC分类号： G10L21/02
CPC分类号： G10L19/26 , G10L21/0208
摘要： A method of filtering a speech signal is presented. The method involves providing a filter (404) suited for reduction of distortion caused by speech coding, estimating acoustic noise in the speech signal, adapting the filter in response to the estimated acoustic noise to obtain an adapted filter, and applying the adapted filter to the speech signal so as to reduce acoustic noise and distortion caused by speech coding in the speech signal.
摘要翻译：提出了一种对语音信号进行滤波的方法。该方法包括提供适合于减少由语音编码引起的失真的滤波器（404），估计语音信号中的声学噪声，响应于所估计的声学噪声来调整滤波器以获得适应的滤波器，以及将适配的滤波器应用于语音信号，以便减少由语音信号中的语音编码引起的声学噪声和失真。

2. 发明授权

US08831959B2 Transform audio codec and methods for encoding and decoding a time segment of an audio signal 有权
标题翻译：转换音频编解码器和用于对音频信号的时间段进行编码和解码的方法
公开(公告)号：US08831959B2
公开(公告)日：2014-09-09
申请号：US13183950
申请日：2011-07-15
申请人： Volodya Grancharov , Sigurdur Sverrisson
发明人： Volodya Grancharov , Sigurdur Sverrisson
IPC分类号： G10L19/00 , H03M5/00 , G10L19/12 , G01L19/02
CPC分类号： G01L19/02 , G10L19/02 , G10L19/0212 , G10L19/032
摘要： Methods and devices for efficient encoding/decoding of a time segment of an audio signal. Methods comprise deriving an indicator, z, of the position in a frequency scale of a residual vector associated with the time segment of the audio signal, and deriving a measure, Φ, related to the amount of structure of the residual vector. The methods further comprise determining whether a predefined criterion involving the measure Φ, the indicator z and a predefined threshold Θ, is fulfilled, which corresponds to estimating whether a change of sign of at least some of the non-zero coefficients of the residual vector would be audible after reconstruction of the audio signal time segment. The amplitude of the coefficients of the residual vector is encoded, and the signs of the coefficients of the residual vector are encoded only when it is determined that the criterion is fulfilled, and thus that a change of sign would be audible.
摘要翻译：用于对音频信号的时间段进行有效编码/解码的方法和装置。方法包括导出与音频信号的时间段相关联的残差矢量的频率范围中的位置的指示符，以及导出与残差向量的结构量有关的测量值Φ。所述方法还包括确定是否满足涉及测量Φ，指示符z和预定义阈值Θ的预定准则，其对应于估计残差向量中的至少一些非零系数的符号变化是否将在音频信号时间段重建后可听见。对残差向量的系数的幅度进行编码，并且仅当确定标准满足时才对编码残差矢量的系数的符号进行编码，从而可以听到符号变化。

3. 发明授权

US08706509B2 Method and a decoder for attenuation of signal regions reconstructed with low accuracy 有权
标题翻译：用于以低精度重建的信号区域衰减的方法和解码器
公开(公告)号：US08706509B2
公开(公告)日：2014-04-22
申请号：US13379054
申请日：2011-12-15
申请人： Sebastian Näslund , Volodya Grancharov , Erik Norvell
发明人： Sebastian Näslund , Volodya Grancharov , Erik Norvell
IPC分类号： G10L19/00 , G10L21/00
CPC分类号： G10L19/02 , G10L19/0212 , G10L19/035 , G10L19/038 , G10L19/10
摘要： The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.
摘要翻译：本发明的实施例通过用允许更积极衰减的自适应衰减方案代替恒定衰减来改进常规衰减方案，而不引入信号频率特性的可听见的改变。

4. 发明授权

US08494864B2 Multi-mode scheme for improved coding of audio 有权
标题翻译：用于改进音频编码的多模式方案
公开(公告)号：US08494864B2
公开(公告)日：2013-07-23
申请号：US12996959
申请日：2008-06-24
申请人： Volodya Grancharov , Stefan Bruhn , Harald Pobloth
发明人： Volodya Grancharov , Stefan Bruhn , Harald Pobloth
IPC分类号： G10L19/00
CPC分类号： G10L19/22
摘要： The present invention relates to an improved scheme for coding of audio. In particular, the present invention relates to an encoder device and a method for coding an input signal in an encoder system. The method comprises applying a first mode to the input signal to form a first output and applying a second mode to the input signal to form a second output. A first processed output is then formed from at least a part of the first output, and a second processed output is formed from at least a part of the second output. Forming a second processed output comprises estimating a part of the input signal from at least a part of the second output. Then, an optimum mode is determined based on the first processed output and the second processed output, and the output according to the optimum mode is selected.
摘要翻译：本发明涉及用于音频编码的改进方案。具体而言，本发明涉及编码器装置和编码器系统中的输入信号编码方法。该方法包括将第一模式应用于输入信号以形成第一输出并将第二模式应用于输入信号以形成第二输出。然后从第一输出的至少一部分形成第一处理输出，并且从第二输出的至少一部分形成第二处理输出。形成第二处理输出包括从第二输出的至少一部分估计输入信号的一部分。然后，基于第一处理输出和第二处理输出确定最佳模式，并且选择根据最佳模式的输出。

5. 发明申请

US20110295599A1 Aligning Scheme for Audio Signals 审中-公开
标题翻译：音频信号对齐方案
公开(公告)号：US20110295599A1
公开(公告)日：2011-12-01
申请号：US13146107
申请日：2009-01-26
申请人： Volodya Grancharov , Anders Ekman
发明人： Volodya Grancharov , Anders Ekman
IPC分类号： G10L19/14 , H03L7/00
CPC分类号： G10L21/00
摘要： Methods, devices, and computer programs described herein may segment a reference signal that corresponds to a non-degraded signal into a plurality of reference signal segments; generate filter coefficients based on each reference signal segment; and filter each reference signal segment with its corresponding generated filter coefficients. The methods, devices, and computer programs may also filter a degraded signal, which includes a delayed signal of the reference signal, with each of the generated filtering coefficients to produce a number of degraded signals equivalent to a number of the plurality of reference signal segments; perform time-wise alignment for each filtered degraded signal with respect to each corresponding filtered reference signal segment; and output a time offset based on the performing.
摘要翻译：本文描述的方法，设备和计算机程序可以将对应于未劣化信号的参考信号分段成多个参考信号段; 基于每个参考信号段产生滤波器系数; 并用其对应的生成的滤波器系数对每个参考信号段进行滤波。方法，装置和计算机程序还可以对生成的滤波系数中的每一个滤波包括参考信号的延迟信号的退化信号，以产生等效于多个参考信号段的数量的多个退化信号 ; 针对每个对应的滤波参考信号段对每个滤波退化信号进行时间对准; 并输出基于执行的时间偏移量。

6. 发明授权

US10121481B2 Post-quantization gain correction in audio coding 有权
公开(公告)号：US10121481B2
公开(公告)日：2018-11-06
申请号：US14002509
申请日：2011-07-04
申请人： Erik Norvell , Volodya Grancharov
发明人： Erik Norvell , Volodya Grancharov
IPC分类号： G10L19/02 , G10L19/032 , G10L19/038 , G10L19/083 , G10L21/0232
摘要： A gain adjustment apparatus (60) for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter (62) configured to estimate an accuracy measure (A(b)) of the shape representation (Ñ(b)), and to determine a gain correction (gc(b)) based on the estimated accuracy measure (A(b)). It also includes an envelope adjuster (64) configured to adjust the gain representation (Ê(b)) based on the determined gain correction.

7. 发明授权

US09546924B2 Transform audio codec and methods for encoding and decoding a time segment of an audio signal 有权
标题翻译：转换音频编解码器和用于对音频信号的时间段进行编码和解码的方法
公开(公告)号：US09546924B2
公开(公告)日：2017-01-17
申请号：US14127761
申请日：2011-06-30
申请人： Volodya Grancharov , Sigurdur Sverrisson
发明人： Volodya Grancharov , Sigurdur Sverrisson
IPC分类号： G10L19/00 , G01L19/02 , G10L19/032 , G10L19/02
CPC分类号： G01L19/02 , G10L19/02 , G10L19/0212 , G10L19/032
摘要： Methods and devices for efficient encoding/decoding of a time segment of an audio signal. The methods comprise deriving an indicator, z, of the position in a frequency scale of a residual vector associated with the time segment of the audio signal, and deriving a measure, Φ, related to the amount of structure of the residual vector. The methods further comprise determining whether a predefined criterion involving the measure Φ, the indicator z and a predefined threshold Θ, is fulfilled, which corresponds to estimating whether a change of sign of at least some of the non-zero coefficients of the residual vector would be audible after reconstruction of the audio signal time segment. The respective amplitude of the coefficients of the residual vector is encoded, and the signs of the coefficients of the residual vector are encoded only when it is determined that the criterion is fulfilled, and thus that a change of sign would be audible.
摘要翻译：用于对音频信号的时间段进行有效编码/解码的方法和装置。所述方法包括导出与音频信号的时间段相关联的残差向量的频率标度中的位置的指示符，以及导出与残差矢量的结构量相关的度量Φ。所述方法还包括确定是否满足涉及测量Φ，指示符z和预定义阈值Θ的预定准则，其对应于估计残差向量中的至少一些非零系数的符号变化是否将在音频信号时间段重建后可听见。编码残差向量的系数的相应幅度，并且仅当确定标准满足时才对编码残差矢量的系数的符号进行编码，并且因此可以听到符号改变。

8. 发明授权

US09424856B2 Filling of non-coded sub-vectors in transform coded audio signals 有权
标题翻译：在变换编码音频信号中填充非编码子矢量
公开(公告)号：US09424856B2
公开(公告)日：2016-08-23
申请号：US14003820
申请日：2011-09-14
申请人： Volodya Grancharov , Sebastian Näslund , Sigurdur Sverrisson
发明人： Volodya Grancharov , Sebastian Näslund , Sigurdur Sverrisson
IPC分类号： G10L19/22 , G10L19/02 , G10L19/028 , G10L19/00
CPC分类号： G10L19/02 , G10L19/0212 , G10L19/028 , G10L19/038 , G10L21/038 , G10L2019/0007
摘要： A spectrum filler for filling non-coded residual sub-vectors of a transform coded audio signal includes a sub-vector compressor (42) configured to compress actually coded residual sub-vectors. A sub-vector rejecter (44) is configured to reject compressed residual sub-vectors that do not fulfill a predetermined sparseness criterion. A sub-vector collector (46) is configured to concatenate the remaining compressed residual sub-vectors to form a first virtual codebook (VC1). A coefficient combiner (48) is configured to combine pairs of coefficients of the first virtual codebook (VC1) to form a second virtual codebook (VC2). A sub-vector filler (50) is configured to fill non-coded residual sub-vectors below a predetermined frequency with coefficients from the first virtual codebook (VC1), and to fill non-coded residual sub-vectors above the predetermined frequency with coefficients from the second virtual codebook (VC2).
摘要翻译：用于填充变换编码音频信号的非编码残余子向量的频谱填充器包括被配置为压缩实际编码的残余子向量的子矢量压缩器（42）。子向量拒绝（44）被配置为拒绝不满足预定稀疏标准的压缩的残余子向量。子矢量收集器（46）被配置为连接剩余的压缩的残余子向量以形成第一虚拟码本（VC1）。系数组合器（48）被配置为组合第一虚拟码本（VC1）的系数对以形成第二虚拟码本（VC2）。子向量填充器（50）被配置为用来自第一虚拟码本（VC1）的系数填充低于预定频率的非编码残差子向量，并且填充具有系数的预定频率以上的非编码残差子向量从第二虚拟码本（VC2）。

9. 发明授权

US09269364B2 Audio encoding/decoding based on an efficient representation of auto-regressive coefficients 有权
标题翻译：基于自回归系数的有效表示的音频编码/解码
公开(公告)号：US09269364B2
公开(公告)日：2016-02-23
申请号：US14355031
申请日：2012-05-15
申请人： Volodya Grancharov , Sigurdur Sverrisson
发明人： Volodya Grancharov , Sigurdur Sverrisson
IPC分类号： G10L25/00 , G10L19/032 , G10L19/06 , G10L21/00 , G10L19/02
CPC分类号： G10L19/038 , G10L19/0204 , G10L19/032 , G10L19/06 , G10L21/038 , G10L2019/001
摘要： 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 (fH) of the parametric spectral representation (f) by weighted averaging based on the quantized elements (fL) flipped around a quantized mirroring frequency (fm), 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.
摘要翻译：描述了用于对部分地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）。编码器包括被配置为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），其被配置为通过基于量化的镜像频率（fm）周围翻转的量化元件（fL）通过加权平均来编码参数频谱表示（f）的高频部分（fH），其分离低频部分与高频部分，以及在闭环搜索过程中从频率格码码本（24）确定的频率网格。还描述了相应的解码器，对应的编码/解码方法和包括这样的编码器/解码器的UE。

10. 发明授权

US09240191B2 Frame based audio signal classification 有权
标题翻译：基于帧的音频信号分类
公开(公告)号：US09240191B2
公开(公告)日：2016-01-19
申请号：US14113616
申请日：2011-04-28
申请人： Volodya Grancharov , Sebastian Näslund
发明人： Volodya Grancharov , Sebastian Näslund
IPC分类号： G10L19/00 , G10L19/02 , G10L25/78 , G10L19/20 , G10L25/51
CPC分类号： G10L19/02 , G10L19/20 , G10L25/51 , G10L25/78 , G10L2025/783
摘要： An audio classifier for frame based audio signal classification includes a feature extractor configured to determine, for each of a predetermined number of consecutive frames, feature measures representing at least the following features: auto correlation, frame signal energy, inter-frame signal energy variation. A feature measure comparator is configured to compare each determined feature measure to at least one corresponding predetermined feature interval. A frame classifier is configured to calculate, for each feature interval, a fraction measure representing the total number of corresponding feature measures that fall within the feature interval, and to classify the latest of the consecutive frames as speech if each fraction measure lies within a corresponding fraction interval, and as non-speech otherwise.
摘要翻译：用于基于帧的音频信号分类的音频分类器包括特征提取器，其被配置为针对预定数量的连续帧中的每一个确定表示至少以下特征的特征量度：自相关，帧信号能量，帧间信号能量变化。特征测量比较器被配置为将每个确定的特征测量值与至少一个对应的预定特征间隔进行比较。帧分类器被配置为针对每个特征间隔计算表示落入特征间隔内的相应特征测量的总数的分数度量，并且将最新的连续帧分类为语音，如果每个分数度量位于相应的分数间隔，以及非语音。

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