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    • 33. 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。
    • 34. 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。
    • 37. 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。
    • 40. 发明授权
    • Dynamic compensation of audio signals for improved perceived spectral imbalances
    • 音频信号的动态补偿,用于改善感知频谱不平衡
    • US09391579B2
    • 2016-07-12
    • US13228272
    • 2011-09-08
    • Martin WalshEdward SteinJean-Marc Jot
    • Martin WalshEdward SteinJean-Marc Jot
    • H03G5/00H03G5/02H03G5/16H03G9/00H03G9/02
    • H03G5/025H03G5/005H03G5/165H03G9/005H03G9/025
    • An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth. The final, compensated audio signal is sent to the playback device for playback.
    • 输入音频信号被均衡以根据预期的听力声压级,特定播放设备的输出能力和收听者的独特听觉特征来形成输出音频信号。 首先根据音频信号的属性和母带声级别来确定预期的听音级别。 根据其能力和任何主音量增益,目标聆听电平用于确定特定播放设备的最佳声压级。 这两个级别用于确定基于与人类听觉知觉相关的数据(标准化或直接测量)来制作单个频率的响度。 在可选地扩展信号带宽之后,基于听力损失数据进一步补偿音频,再次被标准化或直接测量。 最后的补偿音频信号被发送到播放设备进行播放。