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
    • 1. 发明申请
    • METHODS AND APPARATUS FOR DOUBLE-INTEGRATION ORTHOGONAL SPACE TEMPERING
    • 双重积分正交空间温度的方法和装置
    • WO2012174208A1
    • 2012-12-20
    • PCT/US2012/042405
    • 2012-06-14
    • FLORIDA STATE UNIVERSITY RESEARCH FOUNDATION, INC.YANG, WeiZHENG, Lianqing
    • YANG, WeiZHENG, Lianqing
    • G05B21/00
    • G06F19/12G06F19/701G06F19/706
    • The orthogonal space random walk (OSRW) algorithm is generalized to be the orthogonal space tempering (OST) method via the introduction of the orthogonal space sampling temperature. Moreover, a double-integration recursion method is developed to enable practically efficient and robust OST free energy calculations, and the algorithm is augmented by a novel theta-dynamics approach to realize both the uniform sampling of order parameter spaces and rigorous end point constraints. In the present work, the double-integration OST method is employed to perform alchemical free energy simulations, specifically to calculate the free energy difference between benzyl phosphonate and difluorobenzyl phosphonate in aqueous solution, to estimate the solvation free energy of the octanol molecule, and to predict the nontrivial Bamase-Barstar binding affinity change induced by the Bamase N58A.
    • 正交空间随机游动(OSRW)算法通过引入正交空间采样温度被推广为正交空间回火(OST)方法。 此外,开发了一种双积分递归方法,以实现实际有效和鲁棒的OST自由能计算,并且该算法通过新的θ-动力学方法来增强,以实现阶参数空间的均匀采样和严格的端点约束。 在本工作中,采用双重积分OST方法进行铝化物自由能模拟,具体计算水溶液中苄基膦酸酯与二氟苄基膦酸酯之间的自由能差异,估计辛醇分子的溶剂化自由能, 预测由Bamase N58A诱导的非平凡的Bamase-Barstar结合亲和力变化。
    • 2. 发明申请
    • SPEECH ENHANCEMENT
    • 语音增强
    • WO2009035615A1
    • 2009-03-19
    • PCT/US2008/010591
    • 2008-09-10
    • DOLBY LABORATORIES LICENSING CORPORATIONBROWN, C. Phillip
    • BROWN, C. Phillip
    • G10L21/02H04S1/00
    • G10L21/02G10L21/0208
    • A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.
    • 一种用于增强语音的方法包括提取音频信号的中心信道,使中心信道的频谱变平,以及将平坦化语音信道与音频信号进行混合,从而增强音频信号中的任何语音。 还公开了一种用于从具有多个声道的音频信号中提取声音的中心声道的方法,用于使音频信号的频谱变平的方法,以及用于检测音频信号中的语音的方法。 还公开了一种语音增强器,其包括中心声道提取,频谱平滑变换器,语音置信发生器和用于将平坦化语音信道与原始音频信号混合的混合器,该原始音频信号与具有检测到的语音的置信度成比例,从而增强任何语音 在音频信号中。
    • 3. 发明申请
    • IMPROVED RATIO OF SPEECH TO NON-SPEECH AUDIO SUCH AS FOR ELDERLY OR HEARING-IMPAIRED LISTENERS
    • 提高对不听话的听众的声音比例,如老年或听觉障碍的听众
    • WO2008100503A2
    • 2008-08-21
    • PCT/US2008/001841
    • 2008-02-12
    • DOLBY LABORATORIES LICENSING CORPORATIONMUESCH, Hannes
    • MUESCH, Hannes
    • G10L19/14G10L21/02
    • H04R25/356H04R2225/43
    • The invention relates to audio signal processing and speech enhancement. In accordance with one aspect, the invention combines a high-quality audio program that is a mix of speech and non-speech audio with a lower-quality copy of the speech components contained in the audio program for the purpose of generating a high-quality audio program with an increased ratio of speech to non-speech audio such as may benefit the elderly, hearing impaired or other listeners. Aspects of the invention are particularly useful for television and home theater sound, although they may be applicable to other audio and sound applications. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.
    • 本发明涉及音频信号处理和语音增强。 根据一个方面,本发明将语音和非语音音频的混合的高质量音频节目与音频节目中包含的语音组件的低质量副本相结合,以产生高质量 音频节目具有增加的语音与非语音的比例,例如可能使老年人,听力障碍者或其他听众受益。 本发明的各方面对于电视和家庭影院声音特别有用,尽管它们可以适用于其它音频和声音应用。 本发明涉及用于执行这些方法的方法,装置以及存储在计算机可读介质上的软件,用于使计算机执行这些方法。
    • 10. 发明申请
    • IMPROVING TRANSIENT PERFORMANCE OF LOW BIT RATE AUDIO CODING SYSTEMS BY REDUCING PRE-NOISE
    • 通过减少预噪声来改善低比特率音频编码系统的瞬态性能
    • WO2002093560A1
    • 2002-11-21
    • PCT/US2002/012957
    • 2002-04-25
    • DOLBY LABORATORIES LICENSING CORPORATIONCROCKETT, Brett, G.
    • CROCKETT, Brett, G.
    • G10L19/14
    • G10L19/02G10L19/0212G10L19/022G10L19/025
    • Distortion artifacts preceding a signal transient in an audio signal stream processed by a transform-based low-bit-rate audio coding system employing coding blocks are reduced by detecting a transient in the audio signal stream and shifting the temporal relationship of the transient with respect to the coding blocks such that the time duration of the distortion artifacts is reduced. The audio data is time scaled in such a way that the transients are temporally repositioned prior to quantization in a transform-based low-bit-rate audio encoder so as to reduce the amount of pre-noise in the decoded audio signal. Alternatively, or in addition, in a transform-based low-bit-rate audio coding system, a transient in the audio signal stream is detected and a portion of the distortion artifacts are time compressed such that the time duration of the distortion artifacts is reduced.
    • 通过采用编码块的基于变换的低比特率音频编码系统处理的音频信号流中的信号瞬态之前的失真伪影通过检测音频信号流中的瞬态并将瞬态的时间关系相对于 编码块使得失真伪影的持续时间减少。 音频数据以这样的方式进行时间缩放,使得在基于变换的低比特率音频编码器中的量化之前暂时重新定位瞬变,以便减少解码音频信号中的预噪声量。 或者或另外,在基于变换的低比特率音频编码系统中,检测音频信号流中的瞬态,并且一部分失真伪影被时间压缩,使得失真伪影的持续时间减少 。