In speech enhancement, it is generally assumed that if you can update your noise estimate on a frame-by-frame basis, you should achieve the highest level of enhancement performance. However, for many noise types and environmental conditions, it is not necessary to perform an update on a frame-by-frame basis to achieve superior performance if the noise structure does not change rapidly. For applications where compute/memory resources are limited, better overall speech performance could be achieved if a more reasonable update rate is estimated so that available compute/memory resources could be made available to the enhancement algorithm itself. In this study, we propose a framework to model the noise structure with the goal of determining the best update rate required to achieve a given quality for speech enhancement. Speech systems generally develop specialized solutions for noise which are unique to each application (i.e., recognition, speaker ID, enhancement etc.). Here we propose a model to predict the noise update rate required to achieve a given quality for enhancement. We evaluate the algorithm across a corpus of four noise types under different levels of degradation. The error between the mean observed and the mean predicted Itakuta-Saito (IS) values of quality are typically between 0.06 to 1.78 IS for our model selected noise frame update rate of 1 frame every 5 frames using the Log-MMSE enhancement scheme. Finally we consider mobile and resource limited applications where such a framework would be useful.