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    • 4. 发明申请
    • ADAPTIVE ENVIRONMENTAL NOISE COMPENSATION FOR AUDIO PLAYBACK
    • 适用于音频播放的环境噪声补偿
    • US20110251704A1
    • 2011-10-13
    • US13084298
    • 2011-04-11
    • Martin WalshEdward SteinJean-Marc JotJames D. Johnston
    • Martin WalshEdward SteinJean-Marc JotJames D. Johnston
    • G06F17/00
    • H04S7/307G10L21/0232G10L21/0364H04R3/04H04S7/301
    • The present invention counterbalances background noise by applying dynamic equalization. A psychoacoustic model representing the perception of masking effects of background noise relative to a desired foreground soundtrack is used to accurately counterbalance background noise. A microphone samples what the listener is hearing and separates the desired soundtrack from the interfering noise. The signal and noise components are analyzed from a psychoacoustic perspective and the soundtrack is equalized such that the frequencies that were originally masked are unmasked. Subsequently, the listener may hear the soundtrack over the noise. Using this process the EQ can continuously adapt to the background noise level without any interaction from the listener and only when required. When the background noise subsides, the EQ adapts back to its original level and the user does not experience unnecessarily high loudness levels.
    • 本发明通过应用动态均衡来抵消背景噪声。 使用表示背景噪声相对于期望的前景音轨的掩蔽效应的感知的心理声学模型来精确地抵消背景噪声。 麦克风采样听众听到的内容,并将所需的音轨与干扰噪声分开。 信号和噪声分量从心理声学的角度进行分析,并且音轨被均衡,使得最初被屏蔽的频率未被屏蔽。 随后,听众可能会听到声音的声音。 使用这个过程,EQ可以连续适应背景噪声水平,而不需要听众的任何交互,只有在需要的时候。 当背景噪声消失时,EQ适应于其原始水平,并且用户不会遇到不必要的高响度水平。
    • 5. 发明授权
    • Method and apparatus for audio loudness and dynamics matching
    • 用于音频响度和动力学匹配的方法和装置
    • US07848531B1
    • 2010-12-07
    • US10043591
    • 2002-01-09
    • Earl VickersJean-Marc Jot
    • Earl VickersJean-Marc Jot
    • H03G3/00
    • H03G7/002H03G7/007
    • The overall loudness of an audio track is calculated by combining a number of weighted loudness measures for segments of the audio track, where the weight applied to each individual loudness measure is a function of the loudness measure. By comparing the original overall loudness measure to a desired overall loudness measure, a gain can be determined that will adjust the loudness level to the desired value. Also disclosed is a dynamic compression method that analyzes the dynamic characteristics of an audio track and determines appropriate compressor parameters. Additionally, the loudness of a post-compressor audio track can be estimated for any given compressor parameters, thus permitting post-compression loudness matching to be done even if the compression is performed in real-time.
    • 通过组合用于音轨的段的多个加权响度测量来计算音轨的总响度,其中应用于每个单独的响度测量的权重是响度测量的函数。 通过将原始的整体响度测量值与期望的整体响度测量值进行比较,可以确定将将响度水平调节到期望值的增益。 还公开了一种动态压缩方法,其分析音轨的动态特性并确定适当的压缩器参数。 此外,可以针对任何给定的压缩机参数来估计压缩后音频轨道的响度,从而即使实时地进行压缩,也能够进行后压缩响度匹配。
    • 6. 发明申请
    • Distributed Spatial Audio Decoder
    • 分布式空间音频解码器
    • US20090110204A1
    • 2009-04-30
    • US12350047
    • 2009-01-07
    • Martin WALSHJean-Marc JotEdward Stein
    • Martin WALSHJean-Marc JotEdward Stein
    • H04R5/00
    • G10L19/16G10L19/008H04R2205/024H04R2420/07H04S5/005
    • This invention describes a method for decentralized decoding of a multichannel audio signal by broadcasting the original encoded data and distributing the decoding process between a plurality of receiving units. This allows for the design and manufacture of scalable multichannel audio reproduction systems having an arbitrary number of output channels, composed of a plurality of generic decoder and loudspeaker units each generating fewer output channels. With distributed decoding, a manufacturer can use “off-the-shelf” stereo or mono signal processors, digital-to-analog converters and amplifier components in each generic decoding module, thus reducing manufacturing costs and complexity requirements for each module while offering unlimited scalability in the total number of output channels.
    • 本发明描述了一种通过广播原始编码数据并在多个接收单元之间分配解码过程来对多声道音频信号进行分散解码的方法。 这允许设计和制造具有任意数量的输出通道的可扩展多声道音频再现系统,该多路音频再现系统由多个通用解码器和每个产生较少输出通道的扬声器单元组成。 通过分布式解码,制造商可以在每个通用解码模块中使用“现成”的立体声或单声道信号处理器,数模转换器和放大器组件,从而降低每个模块的制造成本和复杂性要求,同时提供无限的可扩展性 在输出通道的总数。
    • 9. 发明授权
    • Environmental reverberation processor
    • 环境混响处理器
    • US06188769B1
    • 2001-02-13
    • US09441141
    • 1999-11-12
    • Jean-Marc JotSam DickerLuke Dahl
    • Jean-Marc JotSam DickerLuke Dahl
    • H03G300
    • H04S7/305H04S3/00H04S3/002
    • A method and apparatus for processing sound sources to simulate environmental effects includes source channel blocks for each source and single reverberation block. The source channel blocks include direct, early reflection, and late reverberation blocks for conditioning the source feeds to include delays, spectral changes, and attenuations depending on the position, orientation and directivity of the sound sources, the position and orientation of the listener, and the position and sound transmission and reflection properties of obstacles and walls in a modeled environment. The outputs of the source channel blocks are combined and provided to single reverberation block generating both the early reflections and the late reverberation for all sound sources.
    • 用于处理声源以模拟环境影响的方法和装置包括每个源的源通道块和单个混响块。 源通道块包括用于调节源馈源的直接,早期反射和后期混响块,以根据声源的位置,取向和方向性,听者的位置和方向,包括延迟,频谱变化和衰减,以及 模拟环境中障碍物和墙壁的位置和声音传播和反射特性。 源通道块的输出被组合并提供给单个混响块,同时产生所有声源的早期反射和后期混响。