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    • 3. 发明申请
    • ADAPTIVELY ENCODING PITCH LAG FOR VOICED SPEECH
    • 适应语音的自适应编码LAG
    • WO2013096875A2
    • 2013-06-27
    • PCT/US2012/071435
    • 2012-12-21
    • HUAWEI TECHNOLOGIES CO., LTD.GAO, Yang
    • GAO, Yang
    • G10L25/90G10L19/09G10L19/18
    • System and method embodiments for dual modes pitch coding are provided. The system and method embodiments are configured to adaptively code pitch lags of a voiced speech signal using one of two pitch coding modes according to a pitch length, stability, or both. The two pitch coding modes include a first pitch coding mode with relatively high precision and reduced dynamic range, and a second pitch coding mode with relatively large dynamic range and reduced precision. The first pitch coding mode is used upon determining that the voiced speech signal has a relatively short or substantially stable pitch. The second pitch coding mode is used upon determining that the voiced speech signal has a relatively long or less stable pitch or is a substantially noisy signal.
    • 提供了用于双模音调编码的系统和方法实施例。 系统和方法实施例被配置为根据间距长度,稳定性或两者来使用两种音调编码模式之一自适应地编码有声语音信号的音调滞后。 两个音调编码模式包括具有相对较高精度和降低的动态范围的第一音调编码模式,以及具有相对大的动态范围和精度降低的第二音调编码模式。 在确定有声语音信号具有相对较短或基本上稳定的音调时,使用第一音调编码模式。 第二音调编码模式在确定有声语音信号具有相对较长或较小的稳定音调或者是基本上噪声的信号时被使用。
    • 6. 发明申请
    • SPECTRUM FLATNESS CONTROL FOR BANDWIDTH EXTENSION
    • 用于带宽延伸的光谱平均控制
    • WO2012012414A1
    • 2012-01-26
    • PCT/US2011/044519
    • 2011-07-19
    • HUAWEI TECHNOLOGIES CO., LTD.GAO, Yang
    • GAO, Yang
    • G10L19/00
    • G10L19/002G10L19/022G10L19/24G10L19/26G10L21/038G10L21/0388G10L25/18
    • IIn accordance with an embodiment, a method of decoding an encoded audio bitstream at a decoder includes receiving the audio bitstream, decoding a low band bitstream (207) of the audio bitstream to get low band coefficients (209) in a frequency domain, and copying a plurality of the low band coefficients to a high frequency band location to generate high band coefficients (213). The method further includes processing the high band coefficients (213) to form processed high band coefficients (214). Processing includes modifying an energy envelope of the high band coefficients (213) by multiplying modification gains to flatten or smooth the high band coefficients (213), and applying a received spectral envelope decoded from the received audio bitstream to the high band coefficients (213). The low band coefficients (209) and the processed high band coefficients (214) are then inverse-transformed to the time domain to obtain a time domain output signal (215).
    • 根据实施例,在解码器处解码编码音频比特流的方法包括接收音频比特流,解码音频比特流的低频带比特流(207)以获得频域中的低频带系数(209),并复制 多个低频带系数到高频带位置以产生高频带系数(213)。 该方法还包括处理高频带系数(213)以形成经处理的高频带系数(214)。 处理包括通过将修改增益相乘来平滑或平滑高频带系数(213)来修改高频带系数(213)的能量包络,以及将从接收到的音频比特流解码的接收频谱包络应用于高频带系数(213) 。 然后将低频带系数(209)和经处理的高频带系数(214)逆变换到时域以获得时域输出信号(215)。
    • 8. 发明申请
    • METHODS FOR RECEIVING DIGITAL AUDIO SIGNAL USING PROCESSOR AND CORRECTING LOST DATA IN DIGITAL AUDIO SIGNAL
    • 使用处理器接收数字音频信号并在数字音频信号中校正丢失数据的方法
    • WO2010127617A1
    • 2010-11-11
    • PCT/CN2010/072451
    • 2010-05-05
    • HUAWEI TECHNOLOGIES CO., LTD.GAO, YangHERVE, TaddeiLEI, Miao
    • GAO, YangHERVE, TaddeiLEI, Miao
    • G10L19/04
    • G10L19/0017G10L19/005
    • A method for receiving a digital audio signal using a processor is provided, and the method includes correcting the digital audio signal from lost data, and the correcting comprises: copying frequency domain coefficients of the digital audio signal from a previous frame, adaptively adding random noise coefficients to the frequency domain coefficients copied, and scaling the random noise coefficients and the frequency domain coefficients copied to form recovered frequency domain coefficients, and producing a corrected audio signal from the recovered frequency domain coefficients, wherein the scaling is controlled with a parameter representing a periodicity or harmonicity of the digital audio signal. A method for correcting missing audio data and another method for receiving a digital audio signal using a processor are also provided.
    • 提供一种使用处理器接收数字音频信号的方法,该方法包括从丢失的数据中校正数字音频信号,校正包括:从前一帧复制数字音频信号的频域系数,自适应地增加随机噪声 对所述频域系数进行复制的系数,以及缩放所复制的随机噪声系数和频域系数以形成恢复的频域系数,以及从所述恢复的频域系数产生校正的音频信号,其中,用表示 数字音频信号的周期性或谐波度。 还提供了一种用于校正丢失的音频数据的方法和用于使用处理器接收数字音频信号的另一种方法。
    • 9. 发明申请
    • ADDING SECOND ENHANCEMENT LAYER TO CELP BASED CORE LAYER
    • 添加第二增强层到基于CELP的核心层
    • WO2010031003A1
    • 2010-03-18
    • PCT/US2009/056860
    • 2009-09-14
    • HUAWEI TECHNOLOGIES CO., LTD.GAO, Yang
    • GAO, Yang
    • G10L19/10
    • G10L19/04G10L19/24
    • In an embodiment, a method of transmitting an input audio signal is disclosed. A first coding error of the input audio signal with a scalable codec having a first enhancement layer is encoded, and a second coding error is encoded using a second enhancement layer after the first enhancement layer. Encoding the second coding error includes coding fine spectrum coefficients of the second coding error to produce coded fine spectrum coefficients, and coding a spectral envelope of the second coding error to produce a coded spectral envelope. The coded fine spectrum coefficients and the coded spectral envelope are transmitted.
    • 在一个实施例中,公开了一种发送输入音频信号的方法。 编码具有具有第一增强层的可扩展编解码器的输入音频信号的第一编码错误,并且在第一增强层之后使用第二增强层对第二编码错误进行编码。 编码第二编码误差包括对第二编码误差的精细频谱系数进行编码以产生编码的精细频谱系数,以及对第二编码误差的频谱包络进行编码以产生经编码的频谱包络。 发送编码的精细频谱系数和编码的频谱包络。