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signal mask
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  “signal mask”译为未确定词的双语例句
     There are three basic types of digitally modulated signal: MASK, MPSK and MFSK.
     数字调制信号分三种基本类型:多进制幅度键控MASK、多进制相移键控MPSK和多进制频移键控MFSK。
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  相似匹配句对
     Signal Processing
     信号处理
短句来源
     Signal Attenuators
     信号衰减器
短句来源
     The Eye of the Mask
     假面之眼
短句来源
     There are three basic types of digitally modulated signal: MASK, MPSK and MFSK.
     数字调制信号分三种基本类型:多进制幅度键控MASK、多进制相移键控MPSK和多进制频移键控MFSK。
短句来源
     A method for speech enhancement based on signal subspace and hearing mask ing effect
     一种基于信号子空间和听觉掩蔽效应的语音增强方法
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  signal mask
The transmitted spectral density of the transmitted signal mask shall fall within the spectral, as shown in Figure 17.
      


A novel audio coding algorithm suitable for real time multimedia applications is presented in this paper. Firstly, audio signals are decomposed based on wavelet packet using this algorithm, and then the masking threshold is calculated in wavelet domain. Finally, according to the signal-mask ratio obtained with psychoacoustic model, dynamic bits allocation, quantization, and encoding to the wavelet coefficients of each subband are performed. The experimental results indicate that when compact...

A novel audio coding algorithm suitable for real time multimedia applications is presented in this paper. Firstly, audio signals are decomposed based on wavelet packet using this algorithm, and then the masking threshold is calculated in wavelet domain. Finally, according to the signal-mask ratio obtained with psychoacoustic model, dynamic bits allocation, quantization, and encoding to the wavelet coefficients of each subband are performed. The experimental results indicate that when compact disc audio is compressed to 64 Kbps utilizing this algorithm, the segmentation signal-noise ratio of the recovery signal is 32.32 dB and with virtually no perceptual degradation in sound quality. Computation of this algorithm is simple and real time encoding can be performed in the Pentium 133 MHz computer with no resort to an expensive and dedicated hardware.

文中提出了一种新的适用于实时多媒体应用领域的音频编码算法.该算法首先对音频信号进行小波包分解,然后在小波域中计算掩蔽阈值,最后根据从心理声学模型得到的信号一掩蔽比来对各子带小波系数进行动态比特分配、量化和编码.实验结果表明该算法将 CD音频信号压缩到 64 Kbps时,恢复信号的分段信噪比为 32.32dB,主观上感觉无失真.该算法计算简单,可在不需任何附加硬件的 Pentium 133 MHz个人计算机上实现实时音频编码.

 
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