Title :
A low cost adaptive transform decoder implementation for high-quality audio
Author :
Davidson, Grant ; Anderson, Wallace ; Lovrich, Al
Author_Institution :
Dolby Labs. Inc., San Francisco, CA, USA
Abstract :
Low-cost, 16-b fixed-point digital signal processing (DSP) chips have traditionally been used in real-time wideband audio implementations due to limited arithmetic precision, which can lead to audible roundoff errors. The authors describe a real-time AC-2 stereo digital audio decoder implementation based on one 16-b TMS320C5x DSP. This is achieved by modifying a conventional inverse fast Fourier transform (FFT) computation, using a form of mixed-precision arithmetic, and exploiting the short instruction cycle time of the DSP. Compared with a 16-b single-precision implementation, a moderate increase in required DSP cycle time is incurred. The results indicate that the dynamic range of the 16-b DSP decoder is currently within 1 to 3dB of that obtained by current high-quality 16-b A/D and D/A converters. A further refinement will produce a dynamic range figure which meets or exceeds that obtained by higher-precision fixed-point ALUs
Keywords :
Hi-Fi equipment; audio signals; digital signal processing chips; encoding; signal processing; 16 bit; AC-2 stereo digital audio decoder; TMS320C5x DSP; adaptive transform decoder; fixed-point DSP chips; high-quality audio; inverse fast Fourier transform; low cost; mixed-precision arithmetic; real-time wideband audio; Costs; Decoding; Digital arithmetic; Digital signal processing; Digital signal processing chips; Dynamic range; Fast Fourier transforms; Fixed-point arithmetic; Roundoff errors; Wideband;
Conference_Titel :
Acoustics, Speech, and Signal Processing, 1992. ICASSP-92., 1992 IEEE International Conference on
Conference_Location :
San Francisco, CA
Print_ISBN :
0-7803-0532-9
DOI :
10.1109/ICASSP.1992.226087