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    • 1. 发明授权
    • Directional audio capture adaptation based on alternative sensory input
    • 基于替代感觉输入的定向音频捕获适应
    • US09197974B1
    • 2015-11-24
    • US13735446
    • 2013-01-07
    • Brian ClarkLudger Solbach
    • Brian ClarkLudger Solbach
    • H04M9/08H04R29/00H04N5/00H04M11/00
    • H04M9/08H04N5/60H04N7/141H04R1/406H04R3/005H04R29/004H04R2410/01H04R2499/11
    • A system for audio processing of an acoustic signal captured by at least one microphone based on analysis of a video signal associated with the acoustic signal is provided. The video signal may be dynamically processed and analyzed in real time by the system to locate one or more sound sources and to determine distances between the sound sources and the at least one microphone. This information may be used to perform selective noise suppression, noise cancellation, or selective adjustment of acoustic signal components or energy levels acoustic signal components such that noise or unrelated acoustic signal components associated with sound sources not associated with located sound sources are suppressed or filtered. The video signal analysis may track sound source movement for dynamic steering of an acoustic beam towards the sound source.
    • 提供了一种用于基于与声信号相关联的视频信号的分析,由至少一个麦克风捕获的声信号进行音频处理的系统。 视频信号可以被系统动态处理和实时分析,以定位一个或多个声源并且确定声源与至少一个麦克风之间的距离。 该信息可以用于执行选择性噪声抑制,噪声消除或声学信号分量或能量水平的声信号分量的选择性调整,从而抑制或滤波与与定位的声源不相关联的声源相关联的噪声或不相关的声信号分量。 视频信号分析可以跟踪用于声束朝向声源的动态转向的声源运动。
    • 4. 发明授权
    • Local social conference calling
    • 当地社会电话会议
    • US09007416B1
    • 2015-04-14
    • US13415535
    • 2012-03-08
    • Carlo MurgiaEric Skup
    • Carlo MurgiaEric Skup
    • H04N7/14H04M3/56
    • H04M3/56
    • Provided are systems and methods for communicating between mobile devices. Such methods involve establishing a communication session between the mobile devices using a local network, such as a Wi-Fi network. The methods also involve transmitting data streams from one mobile device in the group to all other mobile devices using this local network. The data streams may include audio and/or video data generated by the mobile devices. For example, mobile device users may conduct a teleconference through the local network and/or share media. The mobile devices may be interconnected using a star or ring topology, which may depend on distances between participants and/or alarm features that reflect connection losses between the mobile devices. Each mobile device may be equipped with one or more microphones for collecting audio signals from its user. A mobile device may also include an audio processing system for noise suppression and/or echo cancellation.
    • 提供了用于在移动设备之间进行通信的系统和方法。 这样的方法涉及使用诸如Wi-Fi网络的本地网络在移动设备之间建立通信会话。 该方法还涉及使用该本地网络将数据流从该组中的一个移动设备发送到所有其他移动设备。 数据流可以包括由移动设备产生的音频和/或视频数据。 例如,移动设备用户可以通过本地网络和/或共享媒体进行电话会议。 可以使用星形或环形拓扑来互连移动设备,这可以取决于参与者之间的距离和/或反映移动设备之间的连接损失的警报特征。 每个移动设备可以配备一个或多个麦克风,用于从其用户收集音频信号。 移动设备还可以包括用于噪声抑制和/或回波消除的音频处理系统。
    • 8. 发明授权
    • Systems and methods for multi-channel dereverberation
    • 多声道混响的系统和方法
    • US08761410B1
    • 2014-06-24
    • US12963497
    • 2010-12-08
    • Carlos AvendanoCarlo Murgia
    • Carlos AvendanoCarlo Murgia
    • H04B3/20
    • H04M9/085
    • The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.
    • 本技术提供可以克服或基本上减轻与周围声环境的多种和动态特性相关的问题的声信号的鲁棒,高质量的混响​​。 本技术利用从多个麦克风接收的声信号来执行基于声信号之间的相关性来准确地识别混响的多方面分析。 由于麦克风之间的空间距离和存在于周围声环境中的反射路径的变化,可以使用声信号之间的相关性来准确地确定一个或多个声信号的部分是否包含期望的语音或不希望的混响。 然后使用这些相关特性来产生施加到一个或多个接收到的声信号的信号修改以保持语音并减少混响。
    • 9. 发明授权
    • Adaptive noise reduction using level cues
    • 使用级别线索自适应降噪
    • US08718290B2
    • 2014-05-06
    • US12693998
    • 2010-01-26
    • Carlo MurgiaCarlos AvendanoKarim YounesMark EveryYe Jiang
    • Carlo MurgiaCarlos AvendanoKarim YounesMark EveryYe Jiang
    • A61F11/06
    • G10K11/16H04R3/005
    • An audio device having two pairs of microphones for noise suppression. Primary and secondary microphones of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation/suppression. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by the tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) used to determine one or more cluster tracking control signal(s). The ILD based cluster tracking signals are used to control adaptation of null-processing noise suppression modules. A noise cancelled primary acoustic signal and ILD based cluster tracking control signals are used during post filtering to adaptively generate a mask to be applied against a speech estimate signal.
    • 具有用于噪声抑制的两对麦克风的音频设备。 三个麦克风的主麦克风和次麦克风可以彼此紧密地定位,以提供用于实现噪声消除/抑制的声学信号。 第三麦克风可以相对于主麦克风或辅助麦克风在扩展麦克风配置中间隔开,以从由第三麦克风和主麦克风提供的音频信号中导出等级线索。 可以使用来自两个麦克风的信号而不是三个麦克风。 通过用于确定一个或多个簇跟踪控制信号的麦克风间级别差(ILD)来表示级别提示。 基于ILD的群集跟踪信号用于控制空处理噪声抑制模块的自适应。 在后滤波期间使用噪声消除的主声信号和基于ILD的群集跟踪控制信号来自适应地生成针对语音估计信号应用的掩码。
    • 10. 发明授权
    • Low complexity bandwidth expansion of speech
    • 低复杂度带宽扩展语音
    • US08700391B1
    • 2014-04-15
    • US12895254
    • 2010-09-30
    • Carlos AvendanoCarlo MurgiaDana Massie
    • Carlos AvendanoCarlo MurgiaDana Massie
    • G10L21/02
    • G10L21/0388G10L25/18
    • Audio signal bandwidth expansion is performed on a narrow bandwidth signal received from a far end source. The far end source may transmit the signal over the audio communication network. The narrow band signal bandwidth is expanded such that the bandwidth exceeds that of the audio communication network. The signal may be expanded by performing frequency folding on the signal. One or more features are determined for the narrow bandwidth signal, and the expanded signal is modified based on a feature. The feature may be signal band energy slope, narrow band signal energy, or some other feature. The modification may be performed by a shelf filter selected based on the feature. The modified signals are provided for additional processing. In some embodiments, a noise component is added to the narrow band signal prior to folding to create an excitation that reduces the appearance of a fully harmonic signal characteristic.
    • 在从远端源接收的窄带宽信号上执行音频信号带宽扩展。 远端源可以通过音频通信网络发送信号。 窄带信号带宽被扩展,使得带宽超过音频通信网络的带宽。 可以通过对信号执行频率折叠来扩展信号。 针对窄带宽信号确定一个或多个特征,并且基于特征修改扩展信号。 该特征可以是信号带能量斜率,窄带信号能量或一些其他特征。 修改可以由基于特征选择的货架过滤器来执行。 修改的信号被提供用于附加处理。 在一些实施例中,在折叠之前将噪声分量添加到窄带信号以产生减少完全谐波信号特征的出现的激励。