Session Z: SIGNAL PROCESSING FORUM - PART 3
Monday, May 13, 14:00 17:00 h A method is presented to convert 1-bit digital audio signals into an analogue signal with sufficient current and voltage to drive loudspeakers. For this goal a novel non-PWM class D power stage is constructed that performs this function with very low distortion and very high efficiency, without the use of feedback or other analogue processing. Results of the prototype development are detailed. In this paper, a new concept for a high speed, high resolution digital pulse-former is presented. A digital pulse-former basically maps each input data word (sample) into a binary pulse of corresponding width. Such binary pulse-length modulated signals are incorporated in digital class-D amplifiers and PWM applications. In our approach a synchronous digital counter converting the rough part of the input sample is combined with a ring oscillator serving as fine counter. This hybrid configuration yields a drastically increased resolution while maintaining moderate clock rates. Results of a first discrete implementation of this concept are discussed. The development of a fully digital audio power amplifier based on PWM or Sigma Delta technologies still has many unsolved practical problems. The most problematic part of the amplifier system is the switching (class-D) power stage. It is extremely difficult to turn on or off the high power voltage impulses as required for a high performance signal quality (e.g. 16 bit CD quality). To use one-bit Sigma-Delta Modulation (SDM) in digital class-D power amplifiers the effective pulse frequency has to be reduced. This paper contains a review about this problem. It will be shown, how the efficiency of the amplifier is degraded by too high pulse frequencies. Fundamental approaches to create high resolution pulse signals with lower pulse rates around 300 to 500 kHz will be shown. One of these is a controlled generation of the pulse-pattern, like it is done in the Bit-Flipping. Alternative approaches can be found, when the dependency of the generated pulse patterns by the loop filter structure is considered. The use of a parametric array in air as a directional audio loudspeaker has been reported in previous literature and the self-demodulation phenomenon is well-understood to seriously distort the generated audible sound as a result of inter-modulation distortion. We propose to model the nonlinear interaction in air using a second-order Volterra kernel at the Rayleigh distance within which sound intensity and parametric conversion efficiency are assumed to be high. Results first obtained from Burgers equation-based numerical simulations and actual measurements are then used in the nonlinear system identification process. Human auditory does not perceive sound of all frequencies with equal loudness. It is known that a low frequency signal needs to be produced with higher power level to have the same loudness as the middle frequency part. There are two ways to overcome this problem. There are either carried out by boosting the power of the low frequency part or utilizing psychoacoustics effect called the missing fundamental. Parametric arrays usage to generate highly directional audible signal has been reported since few decades ago. However the reproduced signal lacks low frequency content. One reason is the relatively low power level produced by the existing parametric array. Utilizing the non-linearity of air, it is proposed to psycho-acoustically enhance the low frequency perception of a parametric array loudspeaker. |
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