ERROR CONCEALMENT METHOD AND APPARATUS FOR AUDIO SIGNAL AND DECODING METHOD AND APPARATUS FOR AUDIO SIGNAL USING THE SAME
    1.
    发明申请
    ERROR CONCEALMENT METHOD AND APPARATUS FOR AUDIO SIGNAL AND DECODING METHOD AND APPARATUS FOR AUDIO SIGNAL USING THE SAME 审中-公开
    用于音频信号和解码方法的错误接收方法和装置以及使用该信号的音频信号的装置

    公开(公告)号:US20150120309A1

    公开(公告)日:2015-04-30

    申请号:US14589617

    申请日:2015-01-05

    CPC classification number: G10L19/005 G10L19/0212 G10L19/022 H03M13/00

    Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

    Abstract translation: 一种用于音频信号的错误隐藏方法和装置,以及使用错误隐藏方法和装置的用于音频信号的解码方法和装置。 错误隐藏方法包括:当在当前帧中发生错误时,基于预定标准,选择频域中的错误隐藏和时域中的错误隐藏中的一个作为当前帧的错误隐藏方案,选择 重复方案和在频域中的内插方案作为当前帧的错误隐藏方案,当选择频域中的错误隐藏时,基于预定标准,并且使用所选择的方案来隐藏当前帧的错误。

    METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO/SPEECH SIGNAL
    3.
    发明申请
    METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO/SPEECH SIGNAL 审中-公开
    编码和解码音频/语音信号的方法和装置

    公开(公告)号:US20140108008A1

    公开(公告)日:2014-04-17

    申请号:US14132224

    申请日:2013-12-18

    CPC classification number: G10L19/00 G10L19/025

    Abstract: Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.

    Abstract translation: 提供了一种对音频/语音信号进行编码的方法,该方法包括根据输入信号中的攻击位置来确定帧的可变长度,即输入信号的处理单元; 将所述输入信号的每个帧变换为频域并将所述帧划分为多个子频带; 并且如果确定在频域中编码子频带的信号,则对频域中的子频带的信号进行编码,并且如果确定子频带的信号被编码在一个时间 域,将子频带的信号逆变换到时域,并对时域中的逆变换信号进行编码。 根据本发明,可以通过控制时间分辨率和频率分辨率来有效地编码音频/语音信号。

    METHOD AND APPARATUS FOR PROCESSING SPEECH SIGNAL
    6.
    发明申请
    METHOD AND APPARATUS FOR PROCESSING SPEECH SIGNAL 有权
    用于处理语音信号的方法和装置

    公开(公告)号:US20160372135A1

    公开(公告)日:2016-12-22

    申请号:US15181716

    申请日:2016-06-14

    Abstract: An apparatus for processing a speech signal is provided. The apparatus includes a communicator comprising communication circuitry configured to transmit and receive data, an actuator comprising actuation circuitry configured to generate vibration and to output a signal, a formant enhancement filter configured to increase a formant of the speech signal, and a controller comprising processing circuitry configured to control the speech signal to be received through the communicator, to estimate at least one formant frequency from the speech signal based on linear predictive coding (LPC), to estimate a bandwidth of the at least one formant frequency, to determine whether the speech signal is a voiced sound or a voiceless sound, to configure the formant enhancement filter based on the at least one formant frequency, the bandwidth of the at least one formant frequency, characteristics of the determined voiced sound or voiceless sound, and signal delivery characteristics of a human body, to apply the formant enhancement filter to the speech signal, and to control the speech signal to which the formant enhancement filter is applied to be output using the actuator through the human body.

    Abstract translation: 提供了一种处理语音信号的装置。 该装置包括通信器,其包括被配置为发送和接收数据的通信电路,包括被配置为产生振动并输出信号的致动电路的致动器,被配置为增加语音信号的共振峰的共振峰增强滤波器,以及包括处理电路 被配置为控制要通过通信器接收的语音信号,以基于线性预测编码(LPC)从所述语音信号估计至少一个共振峰频率,以估计所述至少一个共振峰频率的带宽,以确定所述语音 信号是有声声音或无声声音,以基于至少一个共振峰频率,至少一个共振峰频率的带宽,所确定的有声声音或无声音的特性以及信号传递特性来配置共振峰增强滤波器 人体,将共振峰增强滤波器应用于语音信号, 并且通过人体使用致动器来控制应用共振峰增强滤波器输出的语音信号。

    METHOD AND APPARATUS FOR QUADRATURE MIRROR FILTERING
    10.
    发明申请
    METHOD AND APPARATUS FOR QUADRATURE MIRROR FILTERING 有权
    方法和装置用于准直镜过滤

    公开(公告)号:US20150120306A1

    公开(公告)日:2015-04-30

    申请号:US14524306

    申请日:2014-10-27

    CPC classification number: G10L19/0204 G06F9/3552 H03H17/0272

    Abstract: A method of performing quadrature mirror filter (QMF) synthesis filtering includes recording new samples corresponding to a current time slot at positions of samples to be discarded in a first array that includes modulated QMF sub-band samples. The method further includes extracting samples from the first array to remove aliasing between adjacent sub-bands, determining filter coefficients corresponding to the extracted samples by using modulo operation, and synthesizing a time domain sample where aliasing is removed by using the extracted samples and the filter coefficients.

    Abstract translation: 执行正交镜像滤波器(QMF)合成滤波的方法包括在包括调制的QMF子带样本的第一阵列中记录与当前时隙对应的待采样样本位置的新样本。 该方法还包括从第一阵列提取样本以消除相邻子带之间的混叠,通过使用模运算确定与提取样本相对应的滤波器系数,以及通过使用提取的样本和滤波器合成去除混叠的时域采样 系数。

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