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    • 2. 发明授权
    • Multi-path trellis coded quantization method and multi-path coded quantizer using the same
    • 多路径网格编码量化方法和使用其的多路径编码量化器
    • US08706481B2
    • 2014-04-22
    • US11608956
    • 2006-12-11
    • Kang Eun LeeEun Mi OhHo Sang SungChang Yong Son
    • Kang Eun LeeEun Mi OhHo Sang SungChang Yong Son
    • G10L19/12
    • G10L19/032
    • A method of multi-path trellis coded quantization (TCQ) usable in a speech coding system, and a quantizer using the method. Specifically the method includes calculating accumulated distortions corresponding to 2N survivor paths, wherein N indicates an integer greater than two, each of the 2N survivor paths is going towards one of nodes at an i th stage of a trellis, and i indicates an integer greater than zero, comparing the accumulated distortions respectively corresponding to the 2N survivor paths to select N paths among the 2N survivor paths, wherein the accumulated distortions corresponding to selected N paths are smaller than the accumulated distortions corresponding to unselected N paths establishing the selected N paths as survivor paths going toward an i+1 th stage, and selecting an optimal path among the 2N survivor paths corresponding to each node of a last stage.
    • 一种可用于语音编码系统的多径网格编码量化(TCQ)的方法,以及使用该方法的量化器。 具体地,该方法包括计算对应于2N个幸存路径的累积失真,其中N表示大于2的整数,2N个幸存路径中的每一个正在网格的第i阶段中的一个节点,i表示大于 比较分别对应于2N个幸存路径的累积失真以选择2N个幸存路径中的N个路径,其中对应于所选N个路径的累积失真小于对应于建立所选择的N个路径的未选择的N个路径的累积失真作为幸存者 路径朝向第i + 1级,并且在对应于最后一级的每个节点的2N个幸存路径中选择最佳路径。
    • 3. 发明授权
    • Method and apparatus for encoding and decoding audio/speech signal
    • 用于对音频/语音信号进行编码和解码的方法和装置
    • US08630863B2
    • 2014-01-14
    • US11872116
    • 2007-10-15
    • Chang-yong SonEun-mi OhJung-hoe KimHo-sang SungKang-eun LeeKi-hyun Choo
    • Chang-yong SonEun-mi OhJung-hoe KimHo-sang SungKang-eun LeeKi-hyun Choo
    • G10L19/00G10L19/02
    • G10L19/00G10L19/025
    • 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.
    • 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。
    • 6. 发明授权
    • Adaptive encoding and decoding with forward linear prediction
    • 具有前向线性预测的自适应编码和解码
    • US08010348B2
    • 2011-08-30
    • US11774664
    • 2007-07-09
    • Chang-yong SonEun-mi OhKi-hyun ChooJung-hoe Kim
    • Chang-yong SonEun-mi OhKi-hyun ChooJung-hoe Kim
    • G10L19/02G10L19/04G10L19/08
    • G10L19/18G10L19/0212G10L19/06G10L19/09G10L19/20G10L25/18
    • An adaptive encoding method includes splitting an input signal into a low-frequency band signal and a high-frequency band signal; performing forward adaptive linear prediction on the low-frequency band signal and thus filtering the low-frequency band signal; selectively performing backward adaptive linear prediction or long-term prediction on the filtered low-frequency band signal according to the analysis result of the low-frequency band signal; transforming the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, into a signal in a frequency domain and quantizing the signal; and encoding the high-frequency band signal using the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, or the quantized signal. Therefore, compression efficiency of both speech and music signals can be enhanced, and a robust compression method can be provided for various audio contents at a low bit rate.
    • 一种自适应编码方法,包括将输入信号分成低频带信号和高频带信号; 对低频带信号执行前向自适应线性预测,从而对低频带信号进行滤波; 根据低频带信号的分析结果,对经滤波的低频带信号选择性地执行反向自适应线性预测或长期预测; 将已经进行了反向自适应线性预测或长期预测的低频带信号变换为频域中的信号并量化信号; 并且使用已经执行了反向自适应线性预测或长期预测的低频带信号或量化信号对高频带信号进行编码。 因此,可以提高语音和音乐信号的压缩效率,并且可以以低比特率为各种音频内容提供鲁棒的压缩方法。