Unvoiced/Voiced Decision for Speech Processing

    公开(公告)号:US20180322895A1

    公开(公告)日:2018-11-08

    申请号:US16040225

    申请日:2018-07-19

    Inventor: Yang Gao

    CPC classification number: G10L25/78 G10L19/22 G10L25/93

    Abstract: A method for speech processing includes determining a first unvoicing parameter for a first subframe of a speech signal, and determining a smoothed unvoicing parameter for the first subframe according to a second unvoicing parameter of a second subframe prior to the first subframe of the speech signal. The first unvoicing parameter is determined according to a periodicity parameter and a spectral tilt parameter. The method further includes computing a difference between the first unvoicing parameter for the first subframe and the smoothed unvoicing parameter for the first subframe and determining a classification of the first subframe using the computed difference as a decision parameter. The classification indicates whether the first subframe is an unvoiced speech signal or not an unvoiced speech signal. Bandwidth extension is performed on the speech signal for the first subframe according to the classification of the first subframe.

    Unvoiced/voiced decision for speech processing

    公开(公告)号:US10043539B2

    公开(公告)日:2018-08-07

    申请号:US15391247

    申请日:2016-12-27

    Inventor: Yang Gao

    CPC classification number: G10L25/78 G10L19/22 G10L25/93

    Abstract: A method for speech processing includes determining an unvoicing parameter for a first frame of a speech signal and determining a smoothed unvoicing parameter for the first frame by weighting the unvoicing parameter of the first frame and a smoothed unvoicing parameter of a second frame. The unvoicing parameter reflects a speech characteristic of the first frame. The smoothed unvoicing parameter of the second frame is weighted less heavily when the smoothed unvoicing parameter of the second frame is greater than the unvoicing parameter of the first frame. The method further includes computing a difference, by a processor, between the unvoicing parameter of the first frame and the smoothed unvoicing parameter of the first frame, and determining a classification of the first frame according to the computed difference. The classification includes unvoiced speech or voiced speech. The first frame is processed in accordance with the classification of the first frame.

    Mobile Device with Near Field Communication Function

    公开(公告)号:US20180109291A1

    公开(公告)日:2018-04-19

    申请号:US15567844

    申请日:2015-05-15

    CPC classification number: H04B5/0037 H04B1/3816 H04B5/00 H04B5/0031

    Abstract: A mobile device with the NFC function includes an NFC chip, multiple SIM card slots, a power supply unit, and an eSE integrated into the NFC chip. One SIM card slot is connected to a first power port on the NFC chip. The power supply unit is connected to a second power port on the NFC chip. When the mobile device performs near field communication, the second power port on the NFC chip is triggered to output a first level signal. Each of the rest SIM card slots is connected to the power supply unit. The eSE is connected to the power supply unit. The power supply unit is configured to supply power to the eSE and the SIM card slot that is connected to the power supply unit, when the first level signal is received.

    Packet loss concealment for speech coding

    公开(公告)号:US09767810B2

    公开(公告)日:2017-09-19

    申请号:US15136968

    申请日:2016-04-24

    Inventor: Yang Gao

    CPC classification number: G10L19/005 G10L19/083 G10L19/09 G10L19/22

    Abstract: A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.

    Adaptive high-pass post-filter
    56.
    发明授权
    Adaptive high-pass post-filter 有权
    自适应高通后置滤波器

    公开(公告)号:US09418671B2

    公开(公告)日:2016-08-16

    申请号:US14459100

    申请日:2014-08-13

    Inventor: Yang Gao

    CPC classification number: G10L19/125 G10L19/26 G10L2019/0011

    Abstract: In accordance with an embodiment of the present invention, a method of speech processing included receiving a coded audio signal having coding noise. The method further includes generating a decoded audio signal from the coded audio signal, and determining a pitch corresponding to the fundamental frequency of the audio signal. The method also includes determining the minimum allowable pitch and determining if the pitch of the audio signal is less than the minimum allowable pitch. If the pitch of the audio signal is less than the minimum allowable pitch, applying an adaptive high pass filter on the decoded audio signal to lower the coding noise at frequencies below the fundamental frequency.

    Abstract translation: 根据本发明的实施例,语音处理方法包括接收具有编码噪声的编码音频信号。 该方法还包括从编码的音频信号产生解码的音频信号,以及确定与音频信号的基频对应的音高。 该方法还包括确定最小可允许间距并确定音频信号的音调是否小于最小可允许间距。 如果音频信号的音调小于最小允许音调,则在解码的音频信号上应用自适应高通滤波器,以降低低于基本频率的频率处的编码噪声。

    Packet loss concealment for speech coding
    57.
    发明授权
    Packet loss concealment for speech coding 有权
    语音编码的丢包隐藏

    公开(公告)号:US09336790B2

    公开(公告)日:2016-05-10

    申请号:US14175195

    申请日:2014-02-07

    Inventor: Yang Gao

    CPC classification number: G10L19/005 G10L19/083 G10L19/09 G10L19/22

    Abstract: A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.

    Abstract translation: 通过仅对语音帧内的第一子帧或前两个子帧限制或减小音调增益来实现减少由于语音分组丢失引起的误差传播的语音编码方法。 该方法用于有声语音类。 将音调周期长度与子帧尺寸进行比较,以决定减小帧内的第一子帧或前两个子帧的音调增益。 通过检查音调滞后是否稳定并且音高增益足够高的帧来决定强音阶。 对于强有声的帧,音调滞后,音调增益可以比其他语音类别更有效地编码。

    Adaptive High-Pass Post-Filter
    58.
    发明申请
    Adaptive High-Pass Post-Filter 有权
    自适应高通后置滤波器

    公开(公告)号:US20150051905A1

    公开(公告)日:2015-02-19

    申请号:US14459100

    申请日:2014-08-13

    Inventor: Yang Gao

    CPC classification number: G10L19/125 G10L19/26 G10L2019/0011

    Abstract: In accordance with an embodiment of the present invention, a method of speech processing included receiving a coded audio signal having coding noise. The method further includes generating a decoded audio signal from the coded audio signal, and determining a pitch corresponding to the fundamental frequency of the audio signal. The method also includes determining the minimum allowable pitch and determining if the pitch of the audio signal is less than the minimum allowable pitch. If the pitch of the audio signal is less than the minimum allowable pitch, applying an adaptive high pass filter on the decoded audio signal to lower the coding noise at frequencies below the fundamental frequency.

    Abstract translation: 根据本发明的实施例,语音处理方法包括接收具有编码噪声的编码音频信号。 该方法还包括从编码的音频信号产生解码的音频信号,以及确定与音频信号的基频对应的音高。 该方法还包括确定最小可允许间距并确定音频信号的音调是否小于最小可允许间距。 如果音频信号的音调小于最小允许音调,则在解码的音频信号上应用自适应高通滤波器,以降低低于基本频率的频率处的编码噪声。

    System and Method for Correcting for Lost Data in a Digital Audio Signal
    59.
    发明申请
    System and Method for Correcting for Lost Data in a Digital Audio Signal 审中-公开
    用于校正数字音频信号中丢失数据的系统和方法

    公开(公告)号:US20140207445A1

    公开(公告)日:2014-07-24

    申请号:US14219773

    申请日:2014-03-19

    CPC classification number: G10L19/0017 G10L19/005

    Abstract: In an embodiment, a method of receiving a digital audio signal, using a processor, includes generating a high band time domain signal; generating low band time domain signal; estimating an energy ratio between the high band and the low band from a last good frame; keeping the energy ratio for following frame-erased frames by applying an energy correction scaling gain to a high band signal segment by segment in the time domain; and combining the low band signal and the high band signal into a final output.

    Abstract translation: 在一个实施例中,使用处理器接收数字音频信号的方法包括产生高频带时域信号; 产生低频时域信号; 从最后的良好帧估计高频带和低频带之间的能量比; 通过在时域中逐段对高频带信号进行能量校正缩放增益来保持随后的帧擦除帧的能量比; 并将低频带信号和高频带信号组合成最终输出。

    SPECTRAL ENVELOPE CODING OF ENERGY ATTACK SIGNAL

    公开(公告)号:US20130317813A1

    公开(公告)日:2013-11-28

    申请号:US13888550

    申请日:2013-05-07

    Inventor: Yang Gao

    Abstract: MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis.

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