Adaptive pulse allocation mechanism for linear-prediction based analysis-by-synthesis coders
    1.
    发明申请
    Adaptive pulse allocation mechanism for linear-prediction based analysis-by-synthesis coders 审中-公开
    基于线性预测的自适应脉冲分配机制

    公开(公告)号:US20060235681A1

    公开(公告)日:2006-10-19

    申请号:US11272773

    申请日:2005-11-15

    IPC分类号: G10L19/12

    CPC分类号: G10L19/12

    摘要: A scheme of allocating variable pulses for each frame is proposed to reduce the bit-rate of LP based AbS coders while maintaining the same speech quality. Since speech signal is not stationary, the required pulse count in a speech coder should be variable frame by frame. In this patent the optimal pulse count allocation is provided based on criterion of perceptual distortion analysis. The method comprises receiving source speech data and generating temporary encoded data according to the source speech data, and synthesized speech data according to the temporary encoded data, and adjusting the fixed codebook pulse allocation in temporary encoded data to a minimum required pulse count according to the perceptual disturbance values between the synthesized speech data and the source speech data, and outputting final encoded data accordingly.

    摘要翻译: 提出了为每个帧分配可变脉冲的方案,以在保持相同语音质量的同时降低基于LP的AbS编码器的比特率。 由于语音信号不稳定,语音编码器中所需的脉冲计数应逐帧变化。 在该专利中,基于感知失真分析的准则提供了最佳脉冲计数分配。 该方法包括接收源语音数据并根据源语音数据产生临时编码数据,并根据临时编码数据合成语音数据,并根据临时编码数据将临时编码数据中的固定码本脉冲分配调整到最小所需脉冲数 合成语音数据和源语音数据之间的感知干扰值,并相应地输出最终编码数据。

    Noise reduction method
    2.
    发明授权
    Noise reduction method 有权
    降噪方法

    公开(公告)号:US07133824B2

    公开(公告)日:2006-11-07

    申请号:US10067274

    申请日:2002-02-07

    IPC分类号: G10L21/02

    摘要: A noise reduction method partitions frequency band into multiple sub-bands and estimates the signal-to-noise ratio (SNR) value for each sub-band. An over-subtraction factor of each sub-band is determined based on the estimated SNR value. Then, the clean speech spectrum estimate is determined by performing spectral over-subtraction on each sub-band, so as to determine the clean speech signal from the estimated clean speech spectrum.

    摘要翻译: 降噪方法将频带划分成多个子带,并估计每个子带的信噪比(SNR)值。 基于估计的SNR值确定每个子带的过减减系数。 然后,通过对每个子带执行频谱过减来确定干净的语音频谱估计,以便从估计的干净的语音频谱中确定干净的语音信号。