会员体验
专利管家(专利管理)
工作空间(专利管理)
风险监控(情报监控)
数据分析(专利分析)
侵权分析(诉讼无效)
联系我们
交流群
官方交流:
QQ群: 891211   
微信请扫码    >>>
现在联系顾问~
热词
    • 3. 发明授权
    • Constrained filter encoding of polyphonic signals
    • 复音信号的约束滤波器编码
    • US07725324B2
    • 2010-05-25
    • US11011764
    • 2004-12-15
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • G10L19/00G10L21/04G10L19/02H04R5/00
    • G10L19/008
    • Signals of different channels are combined into one mono signal. A set of adaptive filters, preferably one for each channel, is derived in a respective filter adaptation unit. When an adaptive filter is applied to the mono signal it reconstructs the signal of the respective channel under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal, in encoded form, and the parameters of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.
    • 不同通道的信号被组合成一个单声道信号。 在相应的滤波器适配单元中导出一组适用于每个信道的自适应滤波器。 当自适应滤波器被应用于单声道信号时,它在感知约束下重构相应频道的信号。 感知约束是增益和/或形状约束。 增益约束允许保持通道之间的相对能量,而形状约束允许通过避免频谱零点的不必要的过滤来获得更多的稳定性。 传输的参数是以编码形式的单声道信号,并且自适应滤波器的参数优选地也被编码。 接收机通过应用自适应滤波器和可能的一些额外的后处理来重构不同信道的信号。
    • 4. 发明申请
    • Constrained filter encoding of polyphonic signals
    • 复音信号的约束滤波器编码
    • US20050160126A1
    • 2005-07-21
    • US11011764
    • 2004-12-15
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • G06F17/10
    • G10L19/008
    • Signals of different channels (C1-CN) are combined into one mono signal (x). A set of adaptive filters, preferably one for each channel (C1-CN), is derived in a respective filter adaptation unit (30:1-30:N). When an adaptive filter is applied to the mono signal (x) it reconstructs the signal of the respective channel (C1-CN) under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels (C1-CN) while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal (x), in encoded form, and the parameters (p1-pN) of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.
    • 不同信道(C 1 -C 3 N N)的信号被组合成一个单声道信号(x)。 在各自的滤波器适配单元(30:1-30:N)中导出一组自适应滤波器,优选地每个信道一个(C 1 -C N N N) 。 当自适应滤波器被应用于单声道信号(x)时,它在感知约束下重构相应信道的信号(C 1 -C 1 -C N N)。 感知约束是增益和/或形状约束。 增益约束允许保持通道之间的相对能量(C 1 -C 3 N N),同时形状约束允许通过避免频谱零点的不必要的过滤而获得更多的稳定性。 所传输的参数是编码形式的单声道信号(x),并且自适应滤波器的参数(p <1> N&gt; N&lt; N&gt;)优选地也被编码。 接收机通过应用自适应滤波器和可能的一些额外的后处理来重构不同信道的信号。
    • 6. 发明授权
    • Fidelity-optimized variable frame length encoding
    • 保真优化可变帧长度编码
    • US07809579B2
    • 2010-10-05
    • US11011765
    • 2004-12-15
    • Stefan BruhnIngemar JohanssonAnisse TalebDaniel Enström
    • Stefan BruhnIngemar JohanssonAnisse TalebDaniel Enström
    • G10L19/00G10L15/00
    • G10L19/008G10L19/022
    • Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.
    • 复音信号用于创建主信号,通常为单声道信号和侧信号。 提供了用于侧信号的多种编码方案。 每个编码方案的特征在于一组不同长度的子帧。 子帧的总长度对应于编码方案的编码帧的长度。 在侧信号上使用的编码方案根据和弦信号的当前信号内容而选择。 在优选实施例中,产生侧残留信号作为侧平衡信号和主信号之间的差以平衡系数进行了缩放。 选择平衡因子以最小化侧残留信号。 优化的侧残留信号和平衡因子被编码并作为表示侧信号的编码参数提供。
    • 7. 发明申请
    • Fidelity-optimized variable frame length encoding
    • 保真优化可变帧长度编码
    • US20050149322A1
    • 2005-07-07
    • US11011765
    • 2004-12-15
    • Stefan BruhnIngemar JohanssonAnisse TalebDaniel Enstrom
    • Stefan BruhnIngemar JohanssonAnisse TalebDaniel Enstrom
    • G10L19/00G10L19/02G10L19/14
    • G10L19/008G10L19/022
    • Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.
    • 复音信号用于创建主信号,通常为单声道信号和侧信号。 提供了用于侧信号的多种编码方案。 每个编码方案的特征在于一组不同长度的子帧。 子帧的总长度对应于编码方案的编码帧的长度。 在侧信号上使用的编码方案根据和弦信号的当前信号内容而选择。 在优选实施例中,产生侧残留信号作为侧平衡信号和主信号之间的差以平衡系数进行了缩放。 选择平衡因子以最小化侧残留信号。 优化的侧残留信号和平衡因子被编码并作为表示侧信号的编码参数提供。
    • 9. 发明授权
    • Optimized fidelity and reduced signaling in multi-channel audio encoding
    • 多声道音频编码中优化的保真度和减少的信令
    • US07822617B2
    • 2010-10-26
    • US11358726
    • 2006-02-22
    • Anisse TalebStefan Andersson
    • Anisse TalebStefan Andersson
    • G10L19/00
    • G10L19/022G10L19/002G10L19/008G10L19/24G10L19/26
    • The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (S1) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.
    • 本发明提供了一种用于对多声道音频信号进行编码的有效技术。 本发明依赖于在第一编码过程中编码(S1)多个信道中的一个或多个的信号表示的原理,以及在第二个基于过滤器的编码过程中编码一个或多个信道的另一个信号表示。 根据本发明的基本思想是选择(S2)对于第二编码处理,i)将整个编码帧的帧分配配置成一组子帧的组合,以及ii)每个子帧的滤波器长度, 帧,根据预定标准。 然后根据所选择的组合,在整个编码帧的每个子帧中对第二信号表示进行编码(S3)。 选择帧分配配置并同时调整每个子帧的滤波器长度的可​​能性提供了附加的自由度,并且通常导致改进的性能。
    • 10. 发明申请
    • Methods and Arrangements for Audio Coding and Decoding
    • 音频编码和解码的方法和布置
    • US20090076830A1
    • 2009-03-19
    • US12281953
    • 2007-03-07
    • Anisse Taleb
    • Anisse Taleb
    • G10L19/00
    • G10L19/06G10L19/04G10L19/24
    • A method for audio coding and decoding comprises primary encoding of a present audio signal sample into an encoded representation (T(n)), and non-causal encoding of a first previous audio signal sample into an encoded enhancement representation (ET(n−N+)). The method further comprises providing of the encoded representations to an end user. At the end user, the method comprises primary decoding of the encoded representation (T*(n)) into a present received audio signal sample, and non-causal decoding of the encoded enhancement representation (ET*(n−N+)) into an enhancement first previous received audio signal sample. The method further comprises improving of a first previous received audio signal sample, corresponding to the first previous audio signal sample, based on the enhancement first previous received audio signal sample. Devices and systems for audio coding and decoding are also presented.
    • 一种用于音频编码和解码的方法包括将当前音频信号样本初级编码为编码表示(T(n)),以及将第一先前音频信号样本的非因果编码成编码增强表示(ET(n-N + ))。 该方法还包括向最终用户提供经编码的表示。 在最终用户中,该方法包括将编码表示(T *(n))的主要解码成当前接收的音频信号样本,并将编码的增强表示(ET *(n-N +))的非因果解码转换为 增强先前接收的音频信号样本。 该方法还包括基于增强的先前接收的音频信号样本来改进对应于第一先前音频信号采样的第一先前接收音频信号采样。 还提供了用于音频编码和解码的设备和系统。