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This paper discusses a real-time implementation of a sampling rate conversion technique between two arbitrary chosen sampling rates. In particular, a method to generate fractionally sampled data from discrete-domain (integer) data samples is analyzed and implemented on the Motorola DSP56001 (24-bit digital signal processor). This real-time implementation can be accomplished with a DSP56001 and 5k of data memory. As an example, the conversion from 44.1 kHz (Compact Disc sampling rate) to 48 kHz (Digital Audio Tape sampling rate) is discussed. The assembly code for the main routine and the filtering method to generate interpolated samples are included.

This paper discusses a real-time implementation of an adaptive acoustic echo canceler for a hands-free cellular (mobile) phone operating inside a car. A 512-tap adaptive-finite-impulse-response (AFIR) filter, which is the implementation of the adaptive least-mean-square (LMS) algorithm with tapped-delay-line (TDL) structure, is utilized for the acoustic echo canceller. A simple double-talk detection algorithm and the normalized convergence parameter for the LMS algorithm are discussed and implemented on the DSP56001, which is also used as a host processor for the DSP56200s, A/D and D/A converters. DSP56001 assembly codes are included as well as experimental results and the schematic diagram for the hardware implementation.

Digital Signal Processing (DSP) techniques are becoming the standard for high quality digital sound on compact disks and digital audio tapes in automotive audio entertainment systems. Incorporating DSP technology to improve the quality of AM (amplitude modulation) and FM (frequency modulation) stereo radio reception assures high fidelity performance for all radio functions. This paper will only deal with the AM stereo problem since the digital FM stereo problem has been resolved and demonstrated already [1]. This paper will discuss Motorola Compatible Quadrature Amplitude Modulation (CQAM) AM stereo decoding and how it can be digitally implemented on a Motorola DSP56001. A complete description of CQAM decoding will be given including a brief overview of how the process can be performed digitally.

This paper discusses the implementation of an adaptive acoustic echo canceler for a hands-free mobile phone operating on a digital cellular channel. Two adaptive structures based on the LMS (least mean square) algorithm are implemented. The adaptive lattice structure, which is particularly known for faster convergence relative to the conventional tapped-delay-line (TDL) structure, is used in the initialization stage. After convergence, the lattice coefficients are converted into the coefficients for the TDL structure which can accommodate a larger number of taps in real-time operation due to of its computational simplicity. The conversion method of the TDL coefficients from the lattice coefficients is derived, and the DSP56001 assembly codes for the lattice and TDL structures as well as the simulation results are included.