High quality audio encoding within 128 kbit/s

R. J. Beaton
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引用次数: 1

Abstract

An algorithm to encode 15-KHz bandwidth audio at 128 kb/s has been developed which has important applications in ISDN, digital audio recording, and the broadcasting industry. The algorithm uses subband ADPCM coding combined with adaptive bit allocation over multiple subbands to achieve high audio quality at a net data rate of 4 b per sample. Subband coders are generally found to perform better than the equivalent full-band coder in subjective tests, even though the full-band coder may achieve a higher SNR rating. This is due to perceptual masking effects which cause the band-limited quantization noise below a certain threshold in each subband to be masked by the signal. Dynamic allocation of bits to subbands on a sample-by-sample basis, together with a simple multiplexing strategy, eliminates the need for side information and makes the algorithm robust in the presence of network bit errors.<>
128 kbit/s以内的高质量音频编码
提出了一种以128kb /s的速度对15khz带宽的音频进行编码的算法,该算法在ISDN、数字音频记录和广播行业中有着重要的应用。该算法采用子带ADPCM编码,结合多子带的自适应位分配,在每个样本的净数据速率为4 b的情况下实现高音频质量。在主观测试中,子带编码器通常比等效的全带编码器表现更好,即使全带编码器可能达到更高的信噪比评级。这是由于感知掩蔽效应导致每个子带中低于一定阈值的带限量化噪声被信号掩蔽。在逐个采样的基础上将比特动态分配到子带,加上简单的多路复用策略,消除了对侧信息的需求,并使算法在存在网络比特错误时具有鲁棒性
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