Transform coding of high quality digital audio at low bit rates-algorithms and implementation

K. Brandenburg, H. Gerhser, D. Seitzer, T. Sporer
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引用次数: 4

Abstract

Low-bit-rate coding schemes for coding high-quality digital audio at bit rates of 64 to 128 kbit/s per channel are described. LC-ATC (low-complexity adaptive transform coding) and OCF (optimum coding in the frequency domain) both use transform coding to get high redundancy removal and high adaptation to the perceptual requirements. OCF uses entropy coding of the spectral values for increased coding efficiency and flexibility of the coder. Real-time implementations using the DSP32(C) and DSP 56000 digital signal processors are presented for both algorithms. Results using listening tests and noise-to-mask ratio are given: CE quality can be achieved at 2 bit/sample. At 64 kbit/s, the quality at 15 kHz bandwidth is still higher than today's FM radio.<>
低比特率下高质量数字音频的转换编码算法与实现
描述了以每通道64 ~ 128 kbit/s的比特率编码高质量数字音频的低比特率编码方案。LC-ATC(低复杂度自适应变换编码)和OCF(频域最优编码)都采用变换编码来实现高冗余去除和对感知要求的高适应。OCF使用谱值的熵编码来提高编码效率和编码器的灵活性。给出了两种算法在DSP 56000和DSP32(C)数字信号处理器上的实时实现。给出了聆听测试和噪声掩模比的结果:在2比特/采样的情况下可以达到CE质量。在64kbit /s下,15khz带宽下的质量仍然高于今天的FM收音机。
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