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A low-delay audio codec using the ADPCM structure (ADPCM = adaptive differential pulse code modulation) in subbands is presented. With the use of 8 subbands a coarse spectral shaping of the coding noise is obtained and the signal delay is approximately 3 ms. The targeted bit rate is in the range from 128 to 176 kbit/s per channel for near transparent audio quality. The codec uses a cosine-modulated filterbank and backward adaptive calculation of the prediction coefficients and quantization scaling factors. The computations are optimized for a real-time implementation on a fixed-point DSP with an almost constant workload over time. A comparison with the Philips Subband Coder (SBC) and the Fraunhofer Ultra Low Delay Codec (ULD) is performed.
Author (s): Keiler, Florian
Affiliation:
THOMSON, Corporate Research, Audio Processing Lab
(See document for exact affiliation information.)
AES Convention: 120
Paper Number:6748
Publication Date:
2006-05-06
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Session subject:
Low Bit-Rate Audio Coding
Permalink: https://aes2.org/publications/elibrary-page/?id=13552
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Keiler, Florian; 2006; Real-Time Subband-ADPCM Low-Delay Audio Coding Approach [PDF]; THOMSON, Corporate Research, Audio Processing Lab; Paper 6748; Available from: https://aes2.org/publications/elibrary-page/?id=13552
Keiler, Florian; Real-Time Subband-ADPCM Low-Delay Audio Coding Approach [PDF]; THOMSON, Corporate Research, Audio Processing Lab; Paper 6748; 2006 Available: https://aes2.org/publications/elibrary-page/?id=13552