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    • 91. 发明申请
    • Method and system for reducing effects of noise producing artifacts in a speech signal
    • 用于减少语音信号中噪声产生伪像的影响的方法和系统
    • US20090070106A1
    • 2009-03-12
    • US12284805
    • 2008-09-24
    • Yang GaoEyal Shlomot
    • Yang GaoEyal Shlomot
    • G10L19/14
    • H03G3/3089G10L21/045H03G3/341
    • There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1−k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.
    • 提供了一种减少由语音解码系统使用的语音信号的静音区域中产生噪声的噪声的影响的方法。 该方法包括获得语音子帧的多个输入样本; 将多个输入样本中的每一个的能级的绝对值求和以产生总输入电平(gain_in); 平滑总输入电平以产生平滑电平(Level_in_sm); 基于总输入电平,平滑电平和频谱倾斜参数,确定语音子帧在静音区域中; 使用k1 *(Level_in_sm / 1024)+(1-k1)定义增益,其中K1是频谱倾斜参数的函数; 以及使用所述增益来修改所述语音子帧的能级。
    • 92. 发明申请
    • Dual-Pulse Excited Linear Prediction For Speech Coding
    • 用于语音编码的双脉冲激励线性预测
    • US20080154586A1
    • 2008-06-26
    • US11942066
    • 2007-11-19
    • Yang Gao
    • Yang Gao
    • G10L19/12G10L11/06
    • G10L19/10
    • The invention proposed a Dual-Pulse Excitation Model; wherein two pulses of each pair of pulses are always adjacent each other. Only one position index for each pair of pulses needs to be sent to the decoder, which saves bits to code all pulse positions. The magnitudes of each pair of pulses have limited number of patterns. Because the two pulses are adjacent each other, each pair of pulses with different magnitudes can produce different high-pass and/or low-pass effect. Since the magnitudes have enough variation, it is possible to assign the candidate positions of each pair of pulses within a small range in order to save the searching complexity.
    • 本发明提出了双脉冲激励模型; 其中每对脉冲的两个脉冲总是彼此相邻。 需要将每对脉冲对的一个位置索引发送到解码器,从而保存位以对所有脉冲位置进行编码。 每对脉冲的幅度数量有限。 由于两个脉冲彼此相邻,所以具有不同幅度的每对脉冲可以产生不同的高通和/或低通效应。 由于幅度具有足够的变化,因此可以将每对脉冲的候选位置分配在小范围内,以便节省搜索的复杂度。
    • 96. 发明授权
    • Pitch determination based on weighting of pitch lag candidates
    • 基于音调滞后候选的加权的音调确定
    • US07266493B2
    • 2007-09-04
    • US11251179
    • 2005-10-13
    • Huan-Yu SuYang Gao
    • Huan-Yu SuYang Gao
    • G10L11/04
    • G10L19/12G10L19/0204G10L19/09G10L19/18G10L19/20G10L25/90G10L2019/0002G10L2019/0016
    • There is provided a method of selecting a pitch lag value from a plurality of pitch lag candidates for coding a speech signal. The method comprises identifying the plurality of pitch lag candidates from a frame of the speech signal using correlation; classifying the speech signal to obtain a voice classification; determining whether one or more of the plurality of pitch lag candidates are in a temporal neighborhood of one or more previous pitch lag values; favoring the one or more of the plurality of pitch lag candidates determined to be in the temporal neighborhood of the one or more previous pitch lag values, by adaptive weighting, over other ones of the plurality of pitch lag candidates; and selecting the pitch lag value based on the voice classification and the one or more of the plurality of pitch lag candidates favored by the adaptive weighting.
    • 提供了一种从用于编码语音信号的多个音调滞后候选中选择音调滞后值的方法。 该方法包括使用相关性从语音信号的帧中识别多个音调滞后候选; 对语音信号进行分类以获得语音分类; 确定所述多个音调滞后候选中的一个或多个是否在一个或多个先前音调滞后值的时间邻域中; 通过对多个音调滞后候选中的其他音调滞后候选,通过自适应加权来确定被确定为处于一个或多个先前音调滞后值的时间邻域中的多个音调滞后候选中的一个或多个; 以及基于所述语音分类和由所述自适应加权优选的所述多个音调滞后候选中的一个或多个来选择所述音调滞后值。
    • 100. 发明授权
    • System for improved use of pitch enhancement with subcodebooks
    • US07117146B2
    • 2006-10-03
    • US09940904
    • 2001-08-27
    • Yang Gao
    • Yang Gao
    • G01L19/00G01L11/04
    • G10L19/265G10L19/08G10L21/0364
    • A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codec are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech. The overall quality of the system is strongly related to the excitation. In order to enhance the excitation, the system contains a fixed codebook comprising several subcodebooks. The invention reveals a way to apply a pitch enhancement efficiently and differently for different subcodebooks without using additional bits. The technique is particularly applicable to selectable mode vocoder (SMV) systems.