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    CONFERENCE AUDIO MANAGEMENT
    36.
    发明申请
    CONFERENCE AUDIO MANAGEMENT 审中-公开
    会议音频管理

    公开(公告)号：US20160295539A1

    公开(公告)日：2016-10-06

    申请号：US15083974

    申请日：2016-03-29

    Applicant: QUALCOMM Incorporated

    Inventor： Venkatraman S. Atti ,  Daniel J. Sinder ,  Nikolai Leung ,  Vivek Rajendran

    IPC: H04W56/00 ,  H04L12/26 ,  H04M3/56 ,  H04L29/06

    CPC classification number: H04W56/0045 ,  H04L43/087 ,  H04L65/608 ,  H04M3/562 ,  H04M3/568 ,  H04R29/007 ,  H04S7/00 ,  H04S2420/01 ,  H04W80/04 ,  H04W88/02

    Abstract: A method for managing audio during a conference includes receiving, at a first buffer of a mobile device, a first audio stream from a first device associated with a first participant of the conference. The method also includes receiving, at a second buffer of the mobile device, a second audio stream from a second device associated with a second participant of the conference. The method further includes generating a control signal at a delay controller of the mobile device. The control signal is provided to the first buffer and to the second buffer to synchronize first buffered audio that is output from the first buffer with second buffered audio that is output from the second buffer.

    Abstract translation: 一种用于在会议期间管理音频的方法包括在移动设备的第一缓冲器处接收来自与会议的第一参与者相关联的第一设备的第一音频流。 该方法还包括在移动设备的第二缓冲器处接收来自与会议的第二参与者相关联的第二设备的第二音频流。 该方法还包括在移动设备的延迟控制器处生成控制信号。 将控制信号提供给第一缓冲器和第二缓冲器，以使从第一缓冲器输出的第一缓冲音频与从第二缓冲器输出的第二缓冲音频同步。

    Systems and methods for controlling an average encoding rate for speech signal encoding
    37.
    发明授权
    Systems and methods for controlling an average encoding rate for speech signal encoding 有权
    用于控制语音信号编码的平均编码速率的系统和方法

    公开(公告)号：US09263054B2

    公开(公告)日：2016-02-16

    申请号：US14015984

    申请日：2013-08-30

    Applicant: QUALCOMM Incorporated

    Inventor： Subasingha Shaminda Subasingha ,  Vivek Rajendran ,  Venkatesh Krishnan ,  Venkatraman Srinivasa Atti

    IPC: G10L19/24 ,  G10L19/12

    CPC classification number: G10L19/12 ,  G10L19/24

    Abstract: A method for controlling an average encoding rate by an electronic device is described. The method includes obtaining a speech signal. The method also includes determining a first average rate. The method further includes determining a first threshold based on the first average rate. The method additionally includes controlling the average encoding rate by determining at least one other threshold based on the first threshold. The method also includes sending an encoded speech signal.

    Abstract translation: 描述了一种通过电子设备来控制平均编码速率的方法。 该方法包括获得语音信号。 该方法还包括确定第一平均速率。 该方法还包括基于第一平均速率确定第一阈值。 该方法还包括通过基于第一阈值确定至少一个其他阈值来控制平均编码率。 该方法还包括发送编码语音信号。

    SYSTEM AND METHOD OF REDUNDANCY BASED PACKET TRANSMISSION ERROR RECOVERY
    38.
    发明申请
    SYSTEM AND METHOD OF REDUNDANCY BASED PACKET TRANSMISSION ERROR RECOVERY 有权
    基于冗余的分组传输错误恢复的系统和方法

    公开(公告)号：US20160036564A1

    公开(公告)日：2016-02-04

    申请号：US14809098

    申请日：2015-07-24

    Applicant: QUALCOMM Incorporated

    Inventor： Venkatesh Krishnan ,  Vivek Rajendran ,  Sandip Sarkar ,  Subasingha Shaminda Subasingha ,  Venkatraman S. Atti

    IPC: H04L1/08 ,  H04L1/00

    CPC classification number: H04L1/08 ,  H04L1/0041 ,  H04L1/0045 ,  H04L1/008 ,  H04L1/1812 ,  H04L1/1835

    Abstract: A device includes a receiver, a buffer, a transmitter, and an analyzer. The receiver is configured to receive a plurality of packets that corresponds to at least a subset of a sequence of packets. Error correction data of a first packet of the plurality of packets includes a partial copy of a second packet of the plurality of packets. The analyzer is configured to determine whether a particular packet of the sequence is missing from the buffer, and to determine whether a partial copy of the particular packet is stored in the buffer. The analyzer is also configured to send, via the transmitter, a retransmit message to a second device based at least in part on determining that the buffer does not store the particular packet and that the buffer does not store the partial copy of the particular packet.

    Abstract translation: 设备包括接收器，缓冲器，发射器和分析器。 接收器被配置为接收对应于分组序列的至少一个子集的多个分组。 多个分组的第一分组的纠错数据包括多个分组中的第二分组的部分副本。 分析器被配置为确定序列的特定分组是否从缓冲器丢失，并且确定特定分组的部分副本是否存储在缓冲器中。 分析器还被配置为至少部分地基于确定缓冲器不存储特定分组并且缓冲器不存储特定分组的部分副本，经由发送器向第二设备发送重传消息。

    TEMPORAL GAIN ADJUSTMENT BASED ON HIGH-BAND SIGNAL CHARACTERISTIC
    39.
    发明申请
    TEMPORAL GAIN ADJUSTMENT BASED ON HIGH-BAND SIGNAL CHARACTERISTIC 有权
    基于高频信号特征的时间增益调整

    公开(公告)号：US20150380007A1

    公开(公告)日：2015-12-31

    申请号：US14731276

    申请日：2015-06-04

    Applicant: QUALCOMM Incorporated

    Inventor： Venkatraman S. Atti ,  Venkatesh Krishnan ,  Vivek Rajendran ,  Venkata Subrahmanyam Chandra Sekhar Chebiyyam ,  Subasingha Shaminda Subasingha

    IPC: G10L19/12

    CPC classification number: G10L19/12 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/06 ,  G10L19/24 ,  G10L21/0224 ,  G10L21/038 ,  G10L25/12 ,  G10L2019/0016

    Abstract: The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold.

    Abstract translation: 本公开提供了用于调整时间增益参数和用于调整线性预测系数的技术。 时间增益参数的值可以基于音频信号的合成高频带部分与音频信号的高频带部分的比较。 如果高频带部分的较高频率范围的信号特征满足第一阈值，则可以调整时间增益参数。 可以基于对LP阶使用第一值的LP增益操作来确定线性预测（LP）增益。 LP增益可以与LP合成滤波器的能级相关联。 如果LP增益满足第二阈值，LP阶可以减小。

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