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    • 11. 发明授权
    • System and method for stereophonic acoustic echo cancellation
    • 立体声回声消除的系统和方法
    • US09094496B2
    • 2015-07-28
    • US12896639
    • 2010-10-01
    • Heinz Teutsch
    • Heinz Teutsch
    • H04R5/00H04M9/08H04R3/00
    • H04M9/082H04R3/002
    • Disclosed herein are systems, methods, and non-transitory computer-readable storage media for stereophonic acoustic echo cancellation. The method includes collecting, at a same time, a first audio sample of an audio source from a first omnidirectional microphone and a second audio sample of the audio source from a second omnidirectional microphone. The method includes delaying the second audio sample by a first amount of time to yield a delayed second audio sample and combining the delayed second audio sample with the first audio sample to produce a first channel, then delaying the first audio sample by a second amount of time to yield a delayed first audio sample and combining the delayed first audio sample with the second audio sample to produce a second channel. Then the method includes outputting the first channel and the second channel as a stereo audio signal of the audio source.
    • 本文公开了用于立体声回声消除的系统,方法和非暂时的计算机可读存储介质。 该方法包括同时从第二全向麦克风收集来自第一全向麦克风的音频源的第一音频样本和来自第二全向麦克风的音频源的第二音频样本。 该方法包括将第二音频样本延迟第一时间量以产生延迟的第二音频采样并将延迟的第二音频采样与第一音频采样相组合以产生第一通道,然后将第一音频采样延迟第二音量 时间以产生延迟的第一音频采样并将延迟的第一音频采样与第二音频采样相结合以产生第二通道。 然后该方法包括输出第一通道和第二通道作为音频源的立体声音频信号。
    • 13. 发明申请
    • Spatial Presentation of Audio at a Telecommunications Terminal
    • 在电信终端的音频空间呈现
    • US20100272249A1
    • 2010-10-28
    • US12427823
    • 2009-04-22
    • Eric John DiethornHeinz Teutsch
    • Eric John DiethornHeinz Teutsch
    • H04M3/42H04R5/00
    • H04M3/56H04R5/02
    • The present invention utilizes pseudo-stereo for the communication of secondary information to the user of a telecommunications terminal, such as a speakerphone. In particular, the terminal utilizes a method for the presentation of secondary information to the user of the terminal in a teleconference call by adjusting the spatial properties of the monaural audio received at the user's terminal. An audio communication is modified so as to appear that the communicated audio is arriving from a particular direction in relation to the user's approximate position, wherein the direction that is assigned to the audio depends on one or more characteristics of the call participant who is originating the audio. Each characteristic of a call participant on a call can comprise, while not being limited to, the customer satisfaction of the call participant, the urgency of a need of the call participant, the group membership of the call participant, or the product ownership of the call participant.
    • 本发明利用伪立体声将辅助信息通信给诸如免提电话的电信终端的用户。 具体地说,终端利用在电话会议呼叫中通过调整在用户终端接收到的单声道音频的空间特性来向终端的用户呈现辅助信息的方法。 修改音频通信,以便看起来所传送的音频相对于用户的大致位置从特定方向到达,其中分配给音频的方向取决于发起呼叫的呼叫参与者的一个或多个特征 音频。 呼叫中的呼叫参与者的每个特征可以包括但不限于呼叫参与者的客户满意度,呼叫参与者的紧急需要,呼叫参与者的组成员资格或呼叫参与者的产品所有权 呼叫参与者
    • 15. 发明申请
    • SYSTEM AND METHOD FOR METHOD FOR IMPROVING SPEECH INTELLIGIBILITY OF VOICE CALLS USING COMMON SPEECH CODECS
    • 使用通用语音编解码器改善语音呼叫的语音智能的系统和方法
    • US20130262128A1
    • 2013-10-03
    • US13430936
    • 2012-03-27
    • Heinz TeutschJohn Cornelius Lynch
    • Heinz TeutschJohn Cornelius Lynch
    • G10L21/00
    • G10L21/0364
    • System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal.
    • 用于提高编码语音的可懂度的系统和方法,所述方法包括:从网络接收编码语音信号; 从编码的语音信号中提取编码的媒体数据流和一个或多个控制数据分组; 解码编码的媒体数据流以产生解码的语音信号; 升高解码语音信号的上频谱部分以产生升高的语音信号; 并输出升压语音信号。 在另一个实施例中,该方法可以包括:接收未编码的语音信号; 处理所述未编码语音信号,其中所述处理包括从所述未编码语音信号生成未编码数据流; 提升未经编码的数据流的高频谱部分以产生增强的语音信号; 对增强的语音信号进行编码以产生编码语音信号; 并输出升压语音信号。
    • 18. 发明授权
    • Second-order adaptive differential microphone array
    • 二阶自适应差分麦克风阵列
    • US06584203B2
    • 2003-06-24
    • US09999298
    • 2001-10-30
    • Gary W. ElkoHeinz Teutsch
    • Gary W. ElkoHeinz Teutsch
    • H04R300
    • H04R1/406H04R3/005H04R29/00H04R29/005H04R2410/01H04R2430/21
    • A second-order adaptive differential microphone array (ADMA) has two first-order elements (e.g., 802 and 804 of FIG. 8), each configured to convert a received audio signal into an electrical signal. The ADMA also has (i) two delay nodes (e.g., 806 and 808) configured to delay the electrical signals from the first-order elements and (ii) two subtraction nodes (e.g., 810 and 812) configured to generate forward-facing and backward-facing cardioid signals based on differences between the electrical signals and the delayed electrical signals. The ADMA also has (i) an amplifier (e.g., 814) configured to amplify the backward-facing cardioid signal by a gain parameter; (ii) a third subtraction node (e.g., 816) configured to generate a difference signal based on a difference between the forward-facing cardioid signal and the amplified backward-facing cardioid signal; and (iii) a lowpass filter (e.g., 818) configured to filter the difference signal from the third subtraction node to generate the output signal for the second-order ADMA. The gain parameter for the amplifier can be adaptively adjusted to move a null in the back half plane of the ADMA to track a moving noise source. In a subband implementation, a different gain parameter can be adaptively adjusted to move a different null in the back half plane to track a different moving noise source for each different frequency subband.
    • 二阶自适应差分麦克风阵列(ADMA)具有两个一阶元件(例如,图8的802和804),每个元件被配置为将接收到的音频信号转换成电信号。 ADMA还具有(i)被配置为延迟来自一阶元件的电信号的两个延迟节点(例如,806和808),以及(ii)被配置为产生向前和向后的两个减法节点(例如,810和812) 基于电信号和延迟的电信号之间的差异的向后的心形信号。 ADMA还具有(i)被配置为通过增益参数放大后向心形信号的放大器(例如814) (ii)第三减法节点(例如,816),被配置为基于前向心形信号和放大的后向心形信号之间的差异产生差分信号; 以及(iii)低通滤波器(例如,818),被配置为对来自第三减法节点的差分信号进行滤波,以产生用于二阶ADMA的输出信号。 可以自适应地调节放大器的增益参数,以在ADMA的后半平面移动零点,以跟踪移动噪声源。 在子带实现中,可以自适应地调整不同的增益参数以在后半平面中移动不同的零,以跟踪每个不同频率子带的不同移动噪声源。