DSP CALIBRATION FOR ACCURATE TIME WAVEFORM RECONSTRUCTION

Eric Rosenfeld, Bradford Sumner
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引用次数: 3

Abstract

DSP waveform synthesizers cause signal distortion because of inherent limitations in their ability to construct continuous lime waveforms from discrete samples. Often this distortion is minimized by increasing the sampling rate of the synthesizer, thereby decreasing the processing bandwidth. This paper presentsa software-based, calibration routine for correcting the generated waveform. This technique uses a software equalizer to estimate and correct for the system response. The equalizer is created using a recursive least-squares (ULS) algorithm and has the form of a transversal, FIR filter. Thisfilter can then be used toprewarp the input waveform sequence. The results of this calibration method are presented in the final section. Tntroduction The problem addressed in this paper is accurate time-waveform generation. The actual output of a synthesizer is a time waveform transformed from a sequence of digital samples. Here, transformation should be understood as an inherent limitation on waveform reconstruction which can be described for all synthesizers. Specifically, this paper will not treat harmonic distortion or noise effects. In fact, this paper will show that the inherent limitations of waveform synthesizers can be successfully modelled as linear effects. There are three important types of transformation which will be discussed: sinex+m-x distortion, spectral images and group delay . Often these problems are solved by simply increasing the synthesizer sampling rate. This paper will present a new DSP-based technique for directly calibrating these sources of distortion without increasing the sampling rate. Paper 36.3 986 This technique is intended for multitone and complex waveform testing. The issues addressed by this paper must also be addressedin singletone testing, but there the solution is simpler. For singletone testing it is only necessary to generate calibration factors for gain and phase at a small number of frequencies, whereas the technique described here attempts to calibrate across an entire frequency band. iVaveform Synthesizers and their Limitations
DSP校准精确的时间波形重建
由于DSP波形合成器在从离散样本构建连续石灰波形的能力上存在固有的局限性,因此会导致信号失真。通常这种失真是通过增加合成器的采样率,从而减少处理带宽最小化。本文提出了一种基于软件的校准程序,用于校正产生的波形。该技术使用软件均衡器来估计和校正系统响应。均衡器是使用递归最小二乘(ULS)算法创建的,具有横向FIR滤波器的形式。然后可以使用该滤波器来翘曲输入波形序列。最后一节给出了这种校准方法的结果。本文研究的问题是精确的时间波形生成。合成器的实际输出是由一系列数字采样变换而成的时间波形。在这里,变换应该被理解为波形重建的固有限制,这可以描述为所有合成器。具体来说,本文将不处理谐波失真或噪声影响。事实上,本文将表明波形合成器的固有局限性可以成功地建模为线性效应。本文将讨论三种重要的变换类型:正弦+m-x畸变、光谱图像和群延迟。通常通过简单地提高合成器采样率来解决这些问题。本文将提出一种新的基于dsp的技术,在不增加采样率的情况下直接校准这些失真源。该技术适用于多音和复杂波形测试。本文所处理的问题也必须在单例测试中处理,但是单例测试的解决方案更简单。对于单点测试,只需要在少数频率上生成增益和相位的校准因子,而这里描述的技术试图在整个频带上进行校准。iVaveform合成器及其局限性
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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